MixBus Platinum v1.2

Technical Reference Manual

1. Introduction

MixBus Platinum is a comprehensive broadcast audio processor designed for professional FM transmission, HD Radio, and digital streaming delivery. It implements a complete 14-stage serial signal chain plus ProcIQ supervisory intelligence, delivering broadcast-grade audio processing for today’s online and broadcast delivery requirements.

The processor engine was engineered to address the specific challenges of modern broadcast environments where source material arrives in widely varying formats—from heavily compressed HE-AAC to pristine lossless PCM—and must be conditioned for consistent, competitive on-air presentation without sacrificing musicality or listener fatigue management.

1.1 Key Features

APRE
Psychoacoustically-modeled codec repair
Dual Compression
2-band or 5-band multiband
Stereo Enhancement
3-band M/S with ratio guard
Three-Mode Shaping
PurestDrive, PurestSaturation, Clipper GX
Four Delivery Modes
Online, HD Radio, FM 75µs, FM 50µs
SmartBand EQ
6+1 band parametric with adaptive HF
BS.1770-4 Metering
K-weighted LUFS compliance
ProcIQ Intelligence
Real-time spectral analysis

1.2 Signal Chain Overview

The MixBus Platinum signal chain consists of 14 serial processing stages, supervised by ProcIQ’s adaptive intelligence layer. Audio flows left-to-right through every enabled stage in the fixed order shown below.

1 APRE
2 ProcIQ
3 Phase Rot.
4 Stereo Enh.
5 AGC
6 SmartBand
7 Comp.
8 Presence
9 De-Esser
10 Limiter
11 Shaping
12 Output
13 Metering
14 Delivery
ProcIQ supervisory layer monitors and adapts stages 4–12
Stage Name Function
1APREPsychoacoustic codec artifact repair
2ProcIQSupervisory adaptive intelligence
3Phase RotatorCrest-factor reduction via allpass phase rotation
4Stereo Enhancer3-band M/S stereo width and imaging
5AGCAutomatic gain control and leveling
6SmartBand EQ6+1 band parametric equalization with adaptive HF
7Compression2-band or 5-band multiband dynamics
8PresencePresence and air enhancement
9De-EsserSibilance detection and reduction
10TruePeak LimiterBrick-wall peak limiting with TruePeak detection
11ShapingFinal waveshaping (PurestDrive / PurestSaturation / Clipper GX)
12OutputOutput level, stereo encoding, and final gain
13MeteringBS.1770-4 K-weighted LUFS metering
14DeliveryOnline, HD Radio, FM 75µs, or FM 50µs output conditioning

1.3 Parameter Smoothing

All continuous parameters in MixBus Platinum are interpolated to prevent audible artifacts during real-time adjustment. The smoothing time varies by parameter category:

Smoothing Time Parameter Category Rationale
80 ms Frequency parameters (crossover, presence, de-esser, decorrelation) Filter recalculation is computationally expensive
60 ms Ratio, blend, and width parameters Moderate smoothing avoids audible modulation
30 ms Gain, threshold, and makeup parameters Must respond quickly to user adjustment
0 ms (instant) Boolean toggles and enum selectors Snap to new value immediately
Configuration

2. Master Settings

Master Settings govern the global operating mode of MixBus Platinum. These parameters affect how every subsequent stage in the signal chain behaves.

2.1 Controls

Parameter Range Default Description
Input Format HE-AAC / AAC-LC / Lossless HE-AAC Classifies source material. Activates APRE and influences adaptive behaviors in presence, de-esser, and clipper stages.
Input Trim ±18 dB 0 dB Adjusts gain entering the processing chain.
Output Trim ±18 dB 0 dB Adjusts final output level.
DC Blocker 5–40 Hz 15 Hz Removes sub-audible DC offset.

2.2 Input Format and Adaptive Behavior

The Input Format selection automatically configures multiple stages to optimally handle the characteristics of the incoming audio. The table below summarizes the adaptive behavior triggered by each format:

Stage HE-AAC AAC-LC Lossless
APRE Full 4-stage Lite 2-stage Bypassed
Presence 1.0× gain 1.07× gain 1.10× gain
Clipper 1.0× drive/mix 0.82× drive/mix 0.65× drive/mix
De-Esser 0 dB offset +2 dB offset +3.5 dB offset
PRO TIP

If the chain sounds brittle on high-quality content, verify Input Format is not set to HE-AAC. If loudness feels weak on low-bitrate material, verify it is not set to Lossless.

Stage 1 of 14

3. APRE

APRE is the first processing stage in the MixBus Platinum signal chain. It applies psychoacoustically-modeled restoration to compensate for the specific artifacts introduced by lossy audio codecs. Unlike conventional input conditioning that applies static equalization, APRE analyzes the incoming signal in real time and applies targeted corrections only where codec damage is detected.

APRE operates automatically based on the Input Format selection. There are no user-adjustable parameters.

3.1 Stage Architecture

1
Transient Micro-Restoration
HE-AAC only
2
SBR Harmonic Stabilizer
HE-AAC only
3
Stereo Coherence Repair
HE-AAC + AAC-LC
4
Micro-Dynamic Density
HE-AAC + AAC-LC

Stage 1: Transient Micro-Restoration HE-AAC only

Low-bitrate HE-AAC encoding smears transient attacks as the codec allocates bits toward tonal reconstruction at the expense of temporal precision. Stage 1 detects transient events in the 1–4 kHz band using a dual-envelope comparator with a 1.5 ms fast attack and an 80 ms slow baseline. When the fast envelope exceeds the slow envelope by a threshold amount, a brief dynamic high-shelf boost of +0.5 to +2 dB is applied, restoring perceived attack sharpness without adding sustained brightness.

PRO TIP

Stage 1 targets attack restoration specifically. It does not add continuous treble boost.

Stage 2: SBR Harmonic Stabilizer HE-AAC only

HE-AAC uses Spectral Band Replication (SBR) to reconstruct frequencies above the codec’s baseband cutoff (typically 6–14 kHz). SBR produces characteristic modulation artifacts where reconstructed harmonics fluctuate in level from frame to frame.

Stage 2 monitors the modulation depth of the SBR region and applies gentle dynamic smoothing to stabilize the fluctuating harmonics. A small amount of band-limited saturation (tanh waveshaping at 3.5% blend in the 5–10 kHz range) adds subtle harmonic density that masks residual SBR artifacts.

Stage 3: Stereo Coherence Repair HE-AAC + AAC-LC

Codec quantization introduces decorrelation between left and right channels, particularly in the high-frequency region above 8 kHz. Stage 3 measures the running cross-correlation of the HF content and gently reduces the decorrelated component while optionally reinforcing subtle mid-band width in the 2–5 kHz region.

Stage 4: Micro-Dynamic Density HE-AAC + AAC-LC

Lossy codecs sacrifice low-level detail to maintain accuracy on louder content. Stage 4 applies very light upward micro-compression (1.1–1.2:1 ratio) within the 300 Hz–4 kHz band, targeting only low-level material below −24 to −16 dBFS. Gain change is capped at 3 dB.

DESIGN NOTE

APRE is deliberately placed before all other processing stages. By repairing codec artifacts at the input, every downstream stage operates on cleaner source material.

Supervisory Intelligence

4. ProcIQ

ProcIQ is the adaptive supervisory intelligence of MixBus Platinum. Unlike conventional broadcast processors that react blindly to level changes, ProcIQ actively listens to your audio, performing real-time spectral analysis across 8 frequency bands on every single sample from 80 Hz through 12 kHz. From this analysis it derives spectral centroid, spectral tilt, spectral flux, zero-crossing rate, spectral flatness, and per-band energy ratios.

Content is classified as Voice, Music, or Mixed through a hysteresis-stabilized state machine. ProcIQ maintains a rolling temporal memory of spectral and stress data. A closed-loop effectiveness tracker continuously monitors whether corrections are reducing stress. All adaptation is constrained by user-defined ceiling limits.

4.1 How ProcIQ Works

ProcIQ operates through four interconnected subsystems:

8-Band Spectral Analyzer

Center frequencies at 80, 160, 400, 1000, 2500, 5000, 8000, and 12000 Hz. Computes spectral tilt and energy distribution across the full audio bandwidth on every sample.

Scene Classification

A hysteresis-stabilized state machine classifies content as Voice, Music, or Mixed. Two classification engines are available: Classic (heuristic rules) and Neural Lite (softmax classifier). ProcIQ applies bounded parameter offsets to AGC, compression, presence, de-essing, stereo, limiter, clipper, and SmartBand stages based on the current scene.

Stress Model

ProcIQ continuously computes three stress metrics:

  • Artifact Stress — Measures processing artifacts such as distortion and pumping
  • Density Stress — Measures over-compression and loss of dynamics
  • Stereo Stress — Measures stereo image instability and excessive width

Effectiveness Tracker

A closed-loop system with a range of 0.3–1.0 that continuously monitors whether ProcIQ’s corrections are actually reducing the measured stress values. When effectiveness drops, ProcIQ automatically scales back its intervention.

4.2 Controls

Parameter Range Default Description
Enabled On / Off Off Enables or disables ProcIQ supervisory intelligence.
Strength 0–100% 60% Global scaling of all ProcIQ parameter offsets.
Speed 120–6000 ms 900 ms Adaptation time constant (log-scaled). Lower values react faster; higher values provide smoother transitions.
Speech Bias 0–100% 50% Biases the scene classifier toward speech detection. Higher values make ProcIQ more likely to classify ambiguous content as voice.
Artifact Guard 0–100% 70% Controls how aggressively ProcIQ reduces processing when artifacts are detected.
Stereo Safety 0–100% 70% Limits ProcIQ’s stereo width adaptation to prevent excessive imaging shifts.
Clip Safety 0–100% 70% Controls how quickly ProcIQ backs off shaping drive when clipping stress rises.
Mode Classic / Neural Lite Classic Classification engine. Classic uses heuristic rules; Neural Lite uses a softmax classifier for higher accuracy.
Tone Max Delta 0–3.0 dB 1.5 dB Maximum tonal offset ProcIQ may apply to EQ and presence stages.
Stereo Max Delta 0–0.35 0.18 Maximum stereo width offset ProcIQ may apply.
Dynamics Max Delta 0–3.0 dB 1.4 dB Maximum dynamics offset ProcIQ may apply to compression and limiting stages.
Verbose Telemetry On / Off On Enables detailed real-time telemetry output for monitoring ProcIQ decisions.

4.3 Telemetry

When Verbose Telemetry is enabled, ProcIQ exposes the following real-time readouts:

Readout Range Description
Scene Voice / Music / Mixed + confidence % Current content classification and classifier confidence.
Artifact Stress 0–100% Measured processing artifact level.
Density Stress 0–100% Measured over-compression and dynamic range loss.
Stereo Stress 0–100% Measured stereo image instability.
Tone Action 0–100% Current tonal adaptation intensity.
Dynamics Action 0–100% Current dynamics adaptation intensity.
Stereo Action 0–100% Current stereo adaptation intensity.
Protection Action 0–100% Current protective pullback intensity.
Effectiveness 0.3–1.0 Closed-loop measure of whether corrections are reducing stress.
Why Reason code Human-readable explanation of ProcIQ’s current primary action.

4.4 When to Enable ProcIQ

ProcIQ is most beneficial in the following scenarios:

  • Mixed-content broadcasts — Stations that alternate between music, voice, and mixed programming benefit from automatic adaptation.
  • Live radio — Unpredictable source material requires real-time processing adjustments.
  • Unattended operation — ProcIQ provides automated oversight when no engineer is monitoring the chain.

For consistent single-genre content where the processing chain has been carefully tuned, ProcIQ can be disabled. At reduced strength (20–40%), ProcIQ provides a useful safety net without significantly altering the established sound signature.

Stage 3 of 14

5. Phase Rotator

The Phase Rotator reduces the crest factor of asymmetric waveforms by applying frequency-dependent phase shifts in the bass region. It uses four cascaded first-order allpass filter stages, all tuned to 200 Hz. Each allpass stage shifts phase without altering the frequency response. The cascaded result produces 360° of total phase rotation at 200 Hz.

All internal processing operates at f64 (64-bit double precision) for maximum numerical accuracy.

5.1 Controls

Parameter Range Default Description
Enabled On / Off On Enables or disables the phase rotator. No dry/wet control is provided—blending an allpass output with the dry signal would create comb filtering.

5.2 Recommended Settings

Use Case Setting Rationale
Voice-heavy formats On Delivers 2–4 dB of crest-factor reduction on asymmetric speech waveforms.
Mixed content On Beneficial for the majority of broadcast and streaming scenarios.
Music-dominant On or Off Try both settings. Some heavily pre-processed music may not benefit.
PRO TIP

Enable the phase rotator before pushing the clipper harder for speech-heavy formats. The crest-factor reduction allows the clipper to operate more transparently at equivalent loudness levels.

Stage 4 of 14

6. Stereo Enhancer

The Stereo Enhancer is a 3-band Mid/Side processor designed to produce wide, engaging stereo presentation that remains safe under downstream limiting, clipping, and FM pre-emphasis.

6.1 Operating Principle

The incoming stereo signal is decoded into Mid (L+R) and Side (L−R) components, then split into three frequency bands:

  • Low band — Split from the Bass Mono frequency (80–350 Hz)
  • Mid band — Between the bass mono and HF enhance crossover points
  • High band — Split from the HF Enhance Crossover (400–8000 Hz)

Each Side band receives an independent width multiplier. A transient detector operates on the high-band Mid channel with dual envelope followers (2 ms attack / 240 ms release), enabling transient-aware stereo enhancement that can widen the image on percussive events while maintaining stability on sustained material.

6.2 Bass Monophonizer

A high-pass filter applied to the Side channel collapses frequencies below the Bass Mono Freq (60–200 Hz, default 120 Hz) to mono. This prevents intermodulation distortion in FM stereo encoders and maintains tight low-frequency imaging for all delivery modes.

6.3 Per-Band Ratio Guard

The ratio guard enforces Side-to-Mid energy limits on a per-band basis to prevent excessive stereo width from causing problems in downstream processing:

  • FM modes — Auto-enforced caps of 0.82 (75µs) and 0.72 (50µs) ensure compatibility with FM stereo encoding.
  • Online / HD Radio — User-configurable via the Side Limit Ratio parameter.

A global guard with configurable attack and release times smoothly reduces Side energy when the ratio exceeds the limit. A Side lookahead buffer (0–3 ms, default 2 ms) allows the guard to anticipate transients.

6.4 Decorrelation Injection

Decorrelation injection creates the perception of enhanced stereo width by introducing phase-shifted Mid content into the Side channel via two cascaded allpass filters. A noise floor detector suppresses injection at approximately −60 dBFS to prevent artificial widening of silence or very quiet passages.

Two algorithm modes are available:

  • Transient Focus — 45% injection, transient-gated. Decorrelation is applied primarily during transient events for a punchy, dynamic stereo image.
  • Decorrelated — Full injection plus 35% HF transient width boost. Provides a consistently wider image across all content.

6.5 Controls

Parameter Range Default Description
Stereo Enabled On / Off On Enables or disables the stereo enhancer stage.
Width 0.0–1.5 1.10 Global stereo width multiplier. Values above 1.0 widen; below 1.0 narrow.
Bass Mono Enabled On / Off On Enables bass mono collapsing below the crossover frequency.
Bass Mono Freq 60–200 Hz 120 Hz Crossover frequency below which Side content is collapsed to mono.
Enhance Amount 0.0–1.0 1.0 Intensity of the stereo enhancement effect.
Transient Sensitivity 0.0–1.0 0.5 Controls how easily the transient detector triggers stereo enhancement boosts.
HF Enhance Crossover 400–2000 Hz 700 Hz Crossover frequency separating the mid and high enhancement bands.
Mid Enhance Ratio 0.0–1.0 0.3 Amount of mid-band stereo enhancement relative to the high band.
Side Limit Enabled On / Off On Enables the per-band ratio guard limiting system.
Side Limit Ratio 0.5–1.5 1.0 Maximum allowed Side-to-Mid energy ratio (Online / HD Radio modes).
Ratio Limit Smooth 5–100 ms 30 ms Smoothing time for ratio guard gain changes.
Pan Law 0 dB / −3 dB −3 dB Pan law compensation for center-panned signals.

6.6 Advanced Controls

Parameter Range Default Description
Algorithm Mode Transient Focus / Decorrelated Transient Focus Selects the decorrelation injection algorithm.
Low Width 0.0–2.0 0.85 Width multiplier for the low frequency band.
Mid Width 0.0–2.0 1.05 Width multiplier for the mid frequency band.
High Width 0.0–2.0 1.10 Width multiplier for the high frequency band.
Transient Boost 0.0–1.0 0.35 Additional width applied during detected transient events.
Decorrelation Amount 0.0–1.0 0.08 Level of phase-shifted Mid content injected into Side.
Decorrelation Freq 200–8000 Hz 2200 Hz Center frequency of the decorrelation allpass filters.
Ratio Low Mul 0.5–2.0 0.80 Per-band ratio guard multiplier for the low band.
Ratio High Mul 0.5–2.0 1.10 Per-band ratio guard multiplier for the high band.
Guard Attack 0.5–50 ms 5 ms Attack time of the ratio guard gain reduction.
Guard Release 10–500 ms 80 ms Release time of the ratio guard gain reduction.
Side Lookahead 0–10 ms 2 ms Lookahead buffer for the ratio guard, allowing transient anticipation.
Stage 5 of 14

7 — Automatic Gain Control (AGC)

The AGC operates as a slow macro-level loudness stabilizer, maintaining consistent output levels across varying source material without audible pumping. It employs a dual-envelope follower architecture: a fast envelope running at 50% of the attack time and 30% of the release time tracks transient peaks for protection, while a slow envelope using the full attack and release values provides the stable reference for gain calculation.

Windowed target-zone gating is a key design feature. When the signal level is already within the configurable Window Size of the target, the AGC release slows dramatically by the Window Release Mult factor. This prevents the leveler from adding unnecessary density to well-controlled material, producing more natural-sounding results that preserve intended dynamic expression.

7.1 — Controls

ParameterRangeDefault
AGC EnabledOn / OffOn
AGC Target−30 to −6 dBFS−14 dBFS
AGC Max Gain0 – 18 dB8 dB
AGC Max Atten0 – 18 dB4 dB
AGC Attack200 – 2000 ms1200 ms
AGC Release500 – 8000 ms4000 ms
AGC DetectorRMS / PeakRMS
AGC LinkOn / OffOn
Window Size0 – 8 dB3 dB
Window Release Mult1 – 20×
PRO TIP If pumping appears, widen the Window Size before slowing release. Too much max gain lifts the noise floor in quiet passages. A high Release Mult (10–20×) makes well-leveled material pass through almost untouched.

7.2 — Crossover & Program Adaptation

The crossover uses LR4 topology (Linkwitz-Riley 4th order) at 24 dB/octave, delivering perfectly flat magnitude response and zero phase difference at the crossover point. The bands recombine after processing without comb filtering or level discontinuities.

Program Adaptation dynamically adjusts the crossover frequency based on spectral energy balance between the low and high bands. When the low band carries significantly more energy, the crossover shifts slightly upward to give more spectral range to the low-band compressor. When the high band dominates, the crossover shifts downward. This adaptation improves consistency across widely varying mixes without overfitting a single fixed split point.

7.3 — Crossover Controls

ParameterRangeDefault
XO EnabledOn / OffOn
XO Freq120 – 300 Hz180 Hz
Program AdaptOn / OffOn
Adapt Strength0.0 – 1.00.45
Stage 6 of 14

8 — SmartBand EQ

SmartBand is a fully-parametric 6-band equalizer with a 7th adaptive band that operates in tandem with ProcIQ’s spectral intelligence. It provides 7 filter shapes: Peaking Bell, Low Shelf, High Shelf, High-Pass, Low-Pass, Band-Pass, and Notch. Full frequency range (20 Hz–20 kHz), gain (±24 dB), and Q (0.1–18.0) are available on every band. The adaptive HF band receives live spectral tilt data — up to +4.5 dB boost on dark content, up to −2.0 dB cut on bright — with a 30% minimum intervention floor. The engine operates at f64 internally.

8.1 — Controls

Global

ParameterRangeDefault
EnabledOn / OffOn
Mix0 – 100%100%

Adaptive HF

ParameterRangeDefault
Adaptive HFOn / OffOn
Adaptive Strength0 – 100%50%
Adaptive Freq2000 – 16000 Hz6000 Hz

Bands 1–6

ParameterRangePer-Band
ShapePeaking Bell / Low Shelf / High Shelf / High-Pass / Low-Pass / Band-Pass / Notch
Frequency20 Hz – 20 kHz
Gain−24 to +24 dB
Q0.1 – 18.0

Default Band Layout

BandShapeFrequency
Band 1High-Pass30 Hz
Band 2Low Shelf100 Hz
Band 3Peaking Bell400 Hz
Band 4Peaking Bell2500 Hz
Band 5High Shelf8000 Hz
Band 6Low-Pass18000 Hz

8.2 — Adaptive HF Settings

When dark content is detected, the adaptive band applies up to +4.5 dB boost. When bright content is detected, up to −2.0 dB cut is applied. The amount is scaled by Adaptive Strength and the ProcIQ effectiveness tracker (30% minimum floor).

PRO TIP Speech / Podcast: 50% Adaptive Strength. Music: 25–35%. FM Broadcast: adjust Band 1 HP to 60–80 Hz.
Stage 7 of 14

9 — Compression

MixBus Platinum provides two selectable compression architectures, switchable with a click-free 10 ms crossfade: Dual-Band mode (LR4 crossover) and 5-Band Multiband mode.

9.1 — Dual-Band Mode

The signal is split at the Crossover frequency (120–300 Hz, default 180 Hz) using an LR4 filter with perfectly flat magnitude response. Each band features a soft-knee algorithm with a 10 dB default knee width: compression begins 5 dB below the threshold and reaches the full ratio 5 dB above it, producing transparent gain control that avoids the abrupt character of hard-knee compression.

Each compressor uses a dual-time-constant release system: a fast release provides quick recovery from transient peaks, while a slow release maintains smooth sustained gain reduction. The Release Blend control sets the static mix between fast and slow. An adaptive algorithm automatically increases the slow release contribution when the signal is far above the threshold, providing program-dependent behavior that sounds natural across widely varying content.

9.2 — Dual-Band Controls

ParameterRangeDefault (Low)Default (High)
Threshold−40 to 0 dBFS−18 dBFS−20 dBFS
Ratio1:1 – 10:11.8:11.8:1
Knee0 – 12 dB10 dB10 dB
Attack1 – 80 ms25 ms12 ms
Release Fast20 – 300 ms120 ms80 ms
Release Slow150 – 2000 ms500 ms400 ms
Release Blend0.0 – 1.00.650.60
Makeup−6 to +12 dB1.5 dB2.0 dB
SC HPF10–120 Hz / 50–400 Hz25 Hz120 Hz
DetectorRMS / PeakRMSRMS

9.3 — Multi-Band Mode

Five bands with split frequencies at 120 / 700 / 2800 / 8000 Hz.

Band Linking

Five unidirectional link paths: B2→B1, B2→B3, B3→B2, B3→B4, B4→B5. When a source band is compressing, its gain reduction pulls the target band in the same direction, scaled by both the global Band Link amount and the per-pair link strength. A GR Spread Limit constrains how far apart adjacent band gain reduction values can diverge.

Low-Level Protection

Low-Level Hold: When input drops below the gate threshold (−70 to −30 dBFS, default −50), all compressor release times are tripled, preventing bands from recovering to unity and revealing the noise floor. Spectral Lock: Below the freeze threshold (−80 to −40 dBFS, default −65), all band gains freeze at their current values, preserving the spectral shape from the previous active audio.

Treble Guard

A fast envelope follower (2 ms attack / 30 ms release) on Band 5 applies extra gain reduction when high-band peak level exceeds a threshold that scales with the Treble Guard setting. This prevents HF harshness under heavy compression without requiring the operator to back off overall ratios.

Stage 8 of 14

10 — Presence

The Presence stage combines a 60% parametric bell with a 40% high-shelf at the target frequency, providing articulation and top-end energy that cuts through broadcast processing without brittleness.

The Adaptive HF system continuously monitors the broadband-to-HF energy ratio of the input signal using 500 ms smoothed envelopes. When the input material is spectrally dull, the system automatically increases presence gain. When material is already bright, it reduces gain. This keeps tonal balance consistent across wildly different masters without manual intervention. An optional Dynamic mode adds HF containment above a configurable threshold.

10.1 — Controls

ParameterRangeDefault
EnabledOn / OffOn
ModeShelf / DynamicShelf
Freq2.5 – 12 kHz6.5 kHz
Gain−3 to +6 dB+2.5 dB
Q0.4 – 1.20.707
Dyn Threshold−24 to −6 dBFS−12 dBFS
Dyn Ratio1:1 – 6:12:1
Dyn Attack1 – 50 ms5 ms
Dyn Release20 – 400 ms120 ms
Adaptive HFOn / OffOn
Adaptive Amount0 – 6 dB3 dB
PRO TIP If highs feel sharp, enable Adaptive HF before reducing gain. Pair presence tuning with de-esser GR monitoring for the best results.
Stage 9 of 14

11 — De-Esser

The De-Esser provides post-presence high-frequency control for sibilance and density management. Placed after the Presence processor and before the limiter, it prevents high-band aggressiveness from the presence stage from being amplified by downstream limiting and clipping. The de-esser operates as a split-band compressor: the signal is split into a low band (passed through unaffected) and a high band (compressed when sibilant energy exceeds the threshold).

The threshold is auto-offset by Input Format: +2 dB for AAC-LC and +3.5 dB for Lossless. This reflects the fact that higher-quality sources tend to have more intact sibilant energy and therefore require less aggressive de-essing to achieve a natural result.

11.1 — Controls

ParameterRangeDefault
EnabledOn / OffOn
Freq3 – 10 kHz6.2 kHz
Threshold−36 to −6 dBFS−18 dBFS
Ratio1:1 – 8:12.2:1
Attack0.5 – 20 ms3 ms
Release20 – 400 ms130 ms
Stage 10 of 14

12 — TruePeak Limiter

Broadcast-grade peak-protection with optional ITU-R BS.1770 true-peak detection. The Master TruePeak Limiting toggle upgrades to 4× oversampled polyphase FIR. The limiter comprises a primary lookahead limiter plus a separate TruePeak Guard.

12.1 — Primary Limiter

Uses a circular delay buffer for true brick-wall limiting at the configured ceiling without overshoots. The configurable Lookahead (0.5–5 ms, default 2.5 ms) determines how far ahead the limiter can see; longer lookahead produces smoother, less audible gain reduction at the cost of slightly more latency.

Program-dependent release extends the release time proportionally to the depth of current gain reduction. When enabled, the effective release is calculated as: release × (1 + (1 − gain) × 2), meaning deeper limiting produces proportionally slower recovery. This makes the limiter less audible on heavily limited material without requiring manual adjustment.

When True Peak Limiting is enabled, the lookahead detector upgrades from sample-rate detection to 4× oversampled polyphase FIR peak detection. A 48-tap Kaiser-windowed sinc filter (β = 7.857, ~80 dB stop-band rejection) is decomposed into twelve-tap polyphase sub-filters evaluating the signal at ¼, ½, and ¾ sample offsets. The lookahead buffer sees these FIR-detected peaks before they arrive and reduces gain smoothly, with zero overshoot at the ceiling.

12.2 — TruePeak Guard

The TruePeak Guard operates as a safety net at the very end of the signal chain, catching any inter-sample peaks that the upstream limiter, clipper, or pre-emphasis stages may have introduced. It uses the same 4× polyphase FIR detection with a fixed 0.15 ms attack when inter-sample peaks exceed the configured ceiling. In FM modes, this stage serves as a post-emphasis overshoot guard, controlling the peak excursions that FM pre-emphasis produces in high-frequency content.

12.3 — Controls

ParameterRangeDefault
Ceiling−6 to 0 dBFS−1 dBFS
Lookahead0.5 – 5 ms2.5 ms
Attack0.1 – 5 ms0.3 ms
Release20 – 500 ms150 ms
Program ReleaseOn / OffOn
Stereo LinkOn / OffOn
Limiter HQOn / OffOff
True Peak GuardOn / OffOff
Final TruePeak GuardOn / OffOn
Final TruePeak Release20 – 400 ms110 ms
TruePeak LimitingOn / OffOff

12.4 — Standards Compliance

The FIR true-peak detection follows the measurement methodology outlined in ITU-R BS.1770 Annex 2. The 48-tap Kaiser-windowed sinc filter (β = 7.857) provides approximately 80 dB of stop-band rejection. The filter is decomposed into four polyphase phases of twelve taps each, enabling efficient 4× oversampled peak detection without running the entire signal path at 4× the sample rate.

Computational overhead: 72 multiply-accumulate operations per stereo sample pair per stage. At 48 kHz this is approximately 3.5 million MACs/second—negligible compared to the rest of the signal chain. When True Peak Limiting is OFF, the limiter and guard fall back to simple sample-rate peak detection with effectively zero additional overhead.

IMPORTANT This is a processing-chain protective limiter, not a certified measurement instrument. Its detection accuracy aligns with the standard methodology used by professional broadcast meters, but for regulatory compliance always verify with a certified ITU-R BS.1770 meter.
Stage 12 of 14

13 — Shaping

Three selectable waveshaping algorithms add harmonic density and character to the processed signal.

PurestDrive
PurestSaturation
Clipper GX

13.1 — PurestDrive

A sin(x) waveshaper with an audio-rate apply factor. Rather than applying a fixed transfer curve to every sample, PurestDrive dynamically calculates its intensity based on the relationship between the current and previous samples. Sustained, body-rich content receives saturation, while attacks and airy high-frequency content pass through with minimal coloring.

The Amount control directly scales the waveshaper intensity from 0 (clean) to 1 (full saturation). The Drive Trim provides level compensation. All processing uses f64 precision (64-bit double) with TPDF dither to maintain low-level accuracy even under heavy saturation.

DESIGN NOTE PurestDrive is based on the Airwindows open-source library and is used under license. The algorithm has been adapted for MixBus Platinum’s f64 signal path.

13.2 — PurestSaturation

A mantissa-preserving polynomial waveshaper:

x − x³/8 + x5/128 − x7/4096 + x9/262144 − x11/33554432

The polynomial uses intentional power-of-two coefficients for mantissa-preserving arithmetic—clean at low input levels, increasingly thick and present harmonics when driven harder.

The Amount control drives the input gain into the waveshaper, scaling from 1× to 10× and determining how deeply the signal enters the nonlinear region. The Character control morphs the output from thick (0.0) to clean (1.0), providing tonal flexibility independent of the drive amount. The Saturation Trim provides output level compensation.

DESIGN NOTE PurestSaturation is based on the Airwindows open-source library and is used under license. The polynomial has been extended for MixBus Platinum’s mantissa-preserving signal path.

13.3 — Clipper GX

A 4-stage broadcast clipping pipeline with 2× oversampling.

Bass Pre-Limiter
Main Waveshaper
Distortion Control
HF Containment
StageDescriptionKey Parameters
1 — Bass Pre-Limiter Catches low-frequency energy below 150 Hz before the main waveshaper. Attack 0.5 ms / Release 15 ms
2 — Main Waveshaper Selectable transfer curves: Tanh, Cubic, Atan. Drive, Mix (dry/wet)
3 — Distortion Control Feedback loop for harmonic management. 0 = inactive, 0.3–0.5 = musical, 0.7+ = clean. Feedback Amount 0.0–1.0
4 — HF Containment Limits harmonic splatter above 4 kHz. Attack 0.3 ms / Release 8 ms

The entire Clipper GX pipeline runs at 2× internal sample rate using midpoint interpolation between adjacent samples, with anti-alias reconstruction filtering (60%/40% blend with 16 kHz lowpass) on the output to minimize aliasing artifacts.

13.4 — FM Path Behavior

When the Output Delivery Mode is set to FM 75µs or FM 50µs, the Shaping stage operates in a special FM mode. A dedicated post-emphasis GX clipping stage processes the pre-emphasized signal, providing the final peak control needed for FM transmission. The FM overshoot guard follows the Final TruePeak Guard toggle in the Limiter tab—when the TruePeak Limiting master toggle is ON, the overshoot guard uses FIR-interpolated peak detection.

Stage 14 of 14

14 — Output & Delivery

14.1 — Delivery Modes

ModeCharacteristic
OnlineFlat — no spectral shaping applied.
HD Radio+4 dB shelf at 3 kHz for enhanced clarity on digital radio.
FM 75 µs+17 dB at 15 kHz. Stereo ratio cap 0.82.
FM 50 µs+14 dB at 15 kHz. Stereo ratio cap 0.72.

14.2 — FM Signal Path

Pre-Emphasis
GX Clipper
Overshoot Guard
Output

The pre-emphasis filter is implemented as a bilinear-transformed analog prototype: H(s) = (1 + s·τ1) / (1 + s·τ2), where τ1 is the emphasis time constant and τ2 = τ1 / 7.94, capping maximum boost at +18 dB. The output is the pre-emphasized signal; de-emphasis is a receiver function and is not included in the transmit path.

IMPORTANT If your FM exciter has its own built-in pre-emphasis, it must be disabled to avoid double-emphasis distortion. MixBus Platinum handles pre-emphasis internally.

14.3 — LUFS Metering

K-weighted momentary loudness measurement per ITU-R BS.1770-4 with 400 ms integration. K-weighting applies a +4 dB high-shelf filter at 1681 Hz followed by a 38 Hz high-pass filter, modeling human hearing sensitivity where bass energy is de-weighted and presence-band energy is emphasized.

ApplicationTargetStandard
Streaming (Spotify / YouTube)−14 LUFS
Streaming (Apple Music)−16 LUFS
Broadcast (Europe)−24 LUFSEBU R128
Broadcast (North America)−24 LUFSATSC A/85

14.4 — TruePeak Metering

Measured in dBTP (decibels relative to True Peak). When the True Peak Limiting master toggle is ON, this meter reflects the 4× FIR oversampled peak level for Online/HD paths using limiter/guard FIR detectors, while FM delivery reports the post-chain sample peak after FM overshoot control. When the master toggle is OFF, the meter reports sample-rate peak levels.

StandardMaximum True Peak
EBU R128≤ −1.0 dBTP
ATSC A/85≤ −2.0 dBTP
Streaming≤ −1.0 dBTP

15 — Technical Specifications

SpecificationValue
Sample Rate48 kHz
Bit Depth32-bit f32 (f64 for phase rotator and Airwindows)
Channels2 (stereo)
Total Parameters205
Signal Chain Stages15
Latency0.5 – 5 ms
CompressionDual-Band or 5-Band Multiband
CrossoverLR4 — 24 dB/oct
LimiterLookahead + 4× FIR TruePeak
ShapingPurestDrive / PurestSaturation / Clipper GX
Delivery ModesOnline / HD Radio / FM 75 µs / FM 50 µs
Loudness StandardITU-R BS.1770-4
FM Pre-EmphasisBilinear analog model, max +18 dB
FM Stereo Safety0.82 (75 µs) / 0.72 (50 µs)
Parameter Smoothing0 / 30 / 60 / 80 ms
PresetsJSON format
TruePeak FIR48-tap Kaiser sinc β=7.857, ~80 dB stop-band

Appendix A — Quick Start Guide

Get up and running with MixBus Platinum in ten steps.

1

Set Input Format

Choose the correct codec for your source material: HE-AAC, AAC-LC, or Lossless. This calibrates all codec-aware offsets throughout the signal chain.

2

Set Output Delivery Mode

Select the target delivery format: Online, HD Radio, FM 75 µs, or FM 50 µs. This configures pre-emphasis, limiter behavior, and stereo safety caps.

3

Adjust Input Trim

Watch the AGC meter while adjusting trim. Aim for the AGC to hover near unity gain with minimal correction.

4

Configure AGC

Set your Target level and Window Size. A wider window and higher Release Mult allow well-leveled content to pass through with minimal intervention.

5

Select Compression Mode

Choose Dual-Band for straightforward dynamic control, or 5-Band Multiband for surgical spectral management. Adjust thresholds and ratios to taste.

6

Adjust Presence with Adaptive HF

Enable Adaptive HF and dial in the presence frequency. The adaptive system will automatically compensate for spectral tilt variations in the source.

7

Configure Stereo Enhancement

Start conservative: Width 1.05–1.15, Decorrelation 0.06–0.12. Check mono compatibility frequently.

8

Set Limiter Ceiling

A ceiling of −1.0 dBFS is the safe starting point for most delivery formats. Enable Final TruePeak Guard for standards compliance.

9

Enable Shaping for Density

Select a shaping algorithm (PurestDrive, PurestSaturation, or Clipper GX) and increase the amount gradually until the desired harmonic density is reached.

10

Monitor Output LUFS

Verify your integrated loudness matches the target for your delivery platform. Fine-tune AGC target and limiter ceiling as needed.

MixBus Platinum Technical Reference Manual · Version 1.2 · February 2026
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