Dynamic Bias Processing

A proprietary designed, program dependent low frequency control stage that runs ahead of the clipping engine. It watches out for your kick and bass relationship with discrete envelope followers; figures out when a transient threatens to exceed the converter’s optimal operating range. The engine then applies targeted gain reduction strictly to that specificied area of spectrum, ensuring intermodulation distortion is minimized without altering the source's natural punch envelope. Making the punch remain apparent while the distortion doesn't.

The engine is crest factor sensitive and identifies whether what's coming through is a transient hit, a sustained bass note, or a decaying tail and reacts accordingly. Set Attack to where the knock feels right and Release to how fast you want the lows to recover.

The Intensity scales the absolute depth of the Dynamic Bias up or down. Set it and forget it, or ride it to taste.

Internal Bias Control

Unlike an ordinary low shelf; this filter operates as a mastering grade Automated Pre-Emphasis / Post-Emphasis network. The Low Shelf Bias attenuates the low-frequency shelf prior to converter stage and automatically applies a reciprocal gain in perfect equilibrium.

The network uses a non-standard implementation of a highly precise Topology-Preserving Transform (TPT) architecture; Using negative values pulls the lows back to tighten the overall response and preserve critical headroom , minimizing low-frequency intermodulation distortion while maintaining absolute equilibrium of the overall linear transfer function. Conversely, applying positive values pushes more low end weight directly into the saturation stage, generating rich bloom and harmonic density.

Converter Emulation & Word Length Reduction

While we believe that the ideal converter should be a "perfect wire" that brings about no change to the signal. Our philosophy to data driven emulation means that we pay attention to the minutia of unintentional mathematical errors and non linearities that occur due to real life limitations and degradations in environmental conditions. So we give you the option to toggle those back on to taste.

Circuit Artifacts is an optional toggle to emulate the harmonics caused by precisely modelled mathematical time errors (jitter), noise and internal component drift artifacts that could be essential to the tone of the hardware converter

SAR Mode switch bypasses modern delta sigma linearity to emulate the Differential Non Linearity (DNL) of early Successive Approximation Register converters. Rather than "texture," this mode accurately models the transfer function errors of hardware, introducing odd order harmonic distortions (turning low level tones into square waves) and signal dependent noise modulation. Subtle at full bit depth, more pronounced as you reduce word length.

Bit depth runs from full Digital 32 bit resolution down to Internal Hardware word lengths such as 20, 21, 22 and 24. A straightforward truncation to shorter word lengths severely deteriorates fine musical detail. The ABC section prevents this. HPDF (Highpass Probability Density Function) or Flat Dither mathematically eliminates truncation distortions and noise modulation. Paired with NS1 through NS5 noise shaping reshapes the frequency content of the quantization errors, shifting quantization noise away from the frequencies you're most sensitive to.

Alias Shaping & DC Suppression

A high order master low pass filter operates during the oversampling and downsampling process to mathematically suppress ultrasonic energy before it can fold back into the audio band as aliasing. Choose Chebyshev for steep, aggressive attenuation of out of band frequencies or Butterworth for a perfectly cleaner, flat amplitude response across the passband. The cutoff and transition width knobs let you tune it to your material.

IIR is zero latency, and is recommended for use in tracking, real time work, or general mixing sessions. FIR is linear phase for maximum transparency in the high end, and is recommended for mastering and critical applications or situations where phase coherence is paramount; latency is reported to the DAW and compensated automatically.

DC removal strips the slow subsonic offset that asymmetric saturation can accumulate. Even a fraction of an LSB of DC offset alters the distortion structure of the signal; So we recommend that you keep it on during any serious clipping session, protecting both peak headroom and speakers.

Stereo Unlinking Divergence

When two channels clip identically, the generated harmonic distortions are phase locked (correlated by dual mono); which collapses and narrows perceived widthof the sound imaging field. Sima breaks that lock by applying a calibrated frequency dependent divergence between the left and right channels, everywhere except the low end.

0% Keeps Stereo Unchanged: Absolute mathematical phase coherence. 100% Introduces a minimal calibrated frequency dependent phase non linearity. 200% Maximized phase divergence and harmonic decorrelation as seen in the analogous to the component tolerance deviations found in discrete dual channel analog hardware.

Crucially, Low frequencies stay phase coherent and intelligently maintained across all settings, guaranteeing absolute mono compatibility and minimizing low end phase cancellations.

Workflow & Controls