MixBus Platinum v1.2

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

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.

1.3 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:

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

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:

3. Adaptive Perceptual Reconstruction Engine (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

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.

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.

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 using a bank of bandpass filters that process every sample from 80 Hz through 12 kHz. Every 128 samples (~2.7 ms), ProcIQ derives spectral centroid, spectral tilt, spectral flux, zero-crossing rate, spectral flatness, and per-band energy ratios from the accumulated filter-bank energy.

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

4.3 Telemetry

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

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.

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

5.2 Recommended Settings

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

6.6 Advanced Controls

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

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

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

Adaptive HF

Bands 1–6

Default Band Layout

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).

9 — Compression

MixBus Platinum provides two selectable compression architectures, switchable with a click-free 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

9.3 — Multi-Band Mode

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

10 — Presence / Exciter

The Presence section offers two selectable modes via the Presence Mode switch: Presence (EQ-based) and Adaptive Exciter (harmonic generation). A single master Enabled toggle bypasses both modes. Switch between them freely — each mode retains its own settings.

10.1 — Presence Mode

The Presence EQ 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.

Presence EQ Controls

10.2 — Adaptive Exciter Mode

The Adaptive Exciter generates new harmonics from existing high-frequency content, adding brightness and air that no EQ can achieve. Unlike simple saturation, it uses an adaptive envelope to normalize input level before waveshaping, producing consistent harmonic density regardless of input dynamics.

Signal Flow

The input signal is split by an LR4 crossover at the Exciter Freq setting. The high-frequency band passes through a dual-speed envelope follower that normalizes its level, then through 2× oversampled Chebyshev waveshaping (using ADAA anti-aliasing) to generate harmonics. A post-filter and DC blocker clean up the result. The shaped harmonics are mixed back with the original dry signal (delayed to compensate for processing latency) according to the Amount control.

Adaptive Exciter Controls

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

12 — TruePeak Limiter

Broadcast-grade brick-wall peak limiter with optional ITU-R BS.1770 TruePeak detection. The TruePeak Limiting toggle upgrades the limiter’s detector to 4× oversampled polyphase FIR, providing precision inter-sample peak control upstream of the shaping stages.

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. Crucially, the attack envelope operates on the incoming sample before it enters the delay buffer, so by the time the audio emerges from the lookahead delay the gain reduction has already converged — guaranteeing true brick-wall protection with zero sample overshoots regardless of the configured attack time.

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 − g) × 2), where g is the linear gain coefficient (0.0 < g ≤ 1.0, with 1.0 = unity). This means 6 dB of gain reduction (g = 0.5) doubles the release time, while deeper limiting (approaching g = 0.0) can triple it — producing proportionally slower recovery on heavily limited material without requiring manual adjustment.

12.2 — TruePeak Limiting

When enabled, the limiter’s peak-detection chain 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 inter-sample peaks before they arrive and reduces gain smoothly, with zero overshoot at the ceiling.

Enabling TruePeak Limiting also forces Limiter HQ on and automatically enables TruePeak Guard in the Output section. This gives you a single switch for full broadcast-grade TruePeak compliance across the entire chain.

Why does this matter when TruePeak Guard already protects the output? The limiter operates upstream of the shaping stages. By detecting inter-sample peaks early, the limiter’s lookahead and program-dependent release can shape gain reduction smoothly and musically — before the signal reaches the clipper and emphasis stages. Without TruePeak Limiting, the downstream TruePeak Guard still catches everything, but it uses a fixed 0.15 ms attack and has no lookahead — so it works harder and can introduce more audible gain pumping on dense material. TruePeak Limiting is precision surgery upstream; TruePeak Guard is the safety net downstream. Together, they deliver the cleanest possible output.

12.3 — Controls

Limiter HQ enables 4× polyphase FIR inter-sample peak detection on the primary limiter’s gain-reduction detector, evaluating peaks at quarter-sample resolution (¼, ½, and ¾ sample offsets) to catch transients that fall between discrete samples. Forced ON when TruePeak Limiting is enabled.

TruePeak Limiting upgrades the limiter to 4× FIR oversampled detection for broadcast-grade inter-sample peak control. Also forces Limiter HQ on and enables TruePeak Guard in the Output section.

TruePeak Guard and TruePeak Release are located in the Output tab — see Section 14.2.

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.

13 — Shaping

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

13.1 — MB Driver

An envelope-driven asymmetric soft saturation stage. Dual-speed envelope followers detect transient peaks versus sustained body—transients pass through cleanly while sustained content receives warm even-order harmonics. The waveshaping transfer function is tanh(k·x) + 0.15·tanh²(k·x), where the tanh² term introduces asymmetric even-harmonic warmth. First-order ADAA (Antiderivative Anti-Aliasing) prevents aliasing without oversampling.

The Drive Amount control (0–100%) sets saturation intensity. At 0% the signal is essentially transparent; at 100% it’s strong saturation. Because the envelope detection modulates the drive depth, the algorithm responds dynamically—sustained notes receive more saturation than transient peaks. The Drive Trim (−12 to +12 dB) provides output makeup gain after the drive stage.

A fast envelope (~0.1 ms attack) catches transient peaks while a slow envelope (~20 ms attack, ~300 ms release) tracks sustained level. The ratio of slow to fast modulates the saturation depth, preserving punch on transients while warming sustained tones.

Best suited for natural analog-console warmth and gentle tape-like saturation where dynamics need to be preserved.

13.2 — MB Saturator

A Chebyshev harmonic sculptor that uses Chebyshev polynomials (T₂ through T₅) to surgically control which harmonics are added to the signal. The Character knob morphs between even harmonics (warm, tape-like) and odd harmonics (gritty, tube-like). Input is soft-limited with tanh before the Chebyshev stage, and ADAA anti-aliasing is applied on the polynomial output.

The Saturation Amount control (0–100%) sets the input drive level, mapping internally to 1×–10× gain into the tanh soft-limiter. The Character control (0–100%) blends the harmonic profile: at 0% the output is mostly 2nd and 4th harmonics (warm, tape-like, even); at 100% it’s mostly 3rd and 5th harmonics (gritty, tube-like, odd); at the default ~30% it produces a balanced warm tone. The Saturation Trim (−12 to +12 dB) provides output makeup gain.

Best suited for tonal shaping and adding specific harmonic color—creative sound design where you want to dial in exactly what type of saturation character you hear.

13.3 — Clipper GX

The Clipper GX introduces a hybrid anti-aliasing architecture to broadcast processing. By combining 2× polyphase FIR oversampling with 1st-order ADAA (Antiderivative Anti-Aliasing) across all nonlinear transfer curves, it achieves master-grade alias suppression without the extreme CPU overhead traditionally required by high-ratio oversampling. The 2× sample rate extends the Nyquist boundary, giving the ADAA math a pristine operating window, while a precision 127-tap equiripple linear-phase half-band decimation filter (>114 dB stopband rejection) surgically removes any residual ultrasonic content before downsampling.

The pipeline operates in four stages:

The Hybrid DSP Approach

Traditional digital clippers rely on brute-force oversampling to push aliasing artifacts above the audible band, which still allows high-frequency distortion to reflect (alias) back down into the midrange. The Clipper GX solves this by calculating the mathematical continuous-time antiderivative of each clipping curve (ADAA).

Operating this ADAA algorithm inside a 2× FIR-oversampled path creates a powerful synergy: the 2× sample rate extends the Nyquist boundary, giving the ADAA math a pristine operating window to suppress aliasing mathematically. Any residual ultrasonic harmonics generated by the waveshaper are then surgically removed by a precision linear-phase half-band decimation filter (127-tap Equiripple, >114 dB stopband rejection, <0.001 dB passband ripple) before downsampling, ensuring master-grade alias suppression with strictly zero phase distortion.

Sonic Benefits

• Zero “Digital Crunch” — By mathematically preventing high-frequency harmonics from folding back into the sensitive 2–6 kHz presence band, the clipper maintains analog-like warmth and clarity even when driven hard.
• Enhanced Transient Punch — Because the aliasing penalty is removed, you can push drum transients and percussive attacks deeper into the waveshaper without generating the brittle harshness associated with standard digital clippers.
• Cleaner Downstream FM Path — A cleaner clipping stage means less high-frequency splatter. For FM delivery modes, this directly translates to a less aggressive workload for the final FM overshoot guard, resulting in a significantly more open and airy top end on the dial.

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 TruePeak Guard toggle in the Output tab — when enabled, the overshoot guard uses 4× FIR-interpolated peak detection for guaranteed inter-sample peak control.

14 — Output & Delivery

14.1 — Delivery Modes

14.2 — TruePeak Guard

Safety limiter positioned at the very end of the signal chain, after all shaping stages (clipper, pre-emphasis). This is the last line of defense — it catches any inter-sample overs that nonlinear processing may have introduced. Always operates with full ITU-R BS.1770 4× FIR oversampled inter-sample peak detection, guaranteeing zero inter-sample overshoots at negligible CPU cost.

Uses a fixed ultra-fast 0.15 ms attack for instantaneous response to peaks and a configurable release (20–400 ms, default 110 ms) for smooth gain recovery. Automatically enabled when TruePeak Limiting is turned on in the Limiter tab.

14.2.1 — Dual-Layer TruePeak Protection

TruePeak Guard and the Limiter’s TruePeak Limiting work as complementary protection circuits:

• TruePeak Limiting (Limiter tab) — upgrades the limiter’s detection upstream, before the signal reaches shaping stages. The limiter’s lookahead and program-dependent release shape gain reduction smoothly and musically. This is precision surgery.
• TruePeak Guard (Output tab) — operates downstream, after all shaping. Catches any new inter-sample peaks that the clipper or emphasis introduced after the limiter. Always uses 4× FIR detection. This is the safety net.

Together, they guarantee broadcast-compliant output with minimal audible artifacts. TruePeak Limiting improves the quality of gain reduction upstream. TruePeak Guard guarantees the output is safe. For maximum protection, enable both.

14.3 — FM Signal Path

The pre-emphasis filter is implemented as a frequency-warped 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. Pre-warping at the zero corner frequency ensures the digital filter accurately matches the analog curve up to 15 kHz, eliminating Nyquist-related frequency cramping. The output is the pre-emphasized signal; de-emphasis is a receiver function and is not included in the transmit path.

14.4 — 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.

14.5 — TruePeak Metering

Measured in dBTP (decibels relative to TruePeak). TruePeak Guard always operates with 4× FIR oversampled detection, so this meter reflects the true inter-sample peak level of the output signal for Online/HD paths. For FM delivery modes, it reports the post-chain peak after FM overshoot control.

15 — Technical Specifications

Appendix A — Quick Start Guide

Get up and running with MixBus Platinum in ten steps.