You just joined a very small group of engineers and producers who own the most comprehensively modeled analog master bus processor ever built. We measured it 2,043 times across nine independent DSP test suites against seven leading commercial plugins. Zero failures, zero warnings. Sphinx 101 doesn't simulate what analog hardware does to your audio - it simulates the actual electronic components inside the hardware and lets the sound emerge from the physics. Welcome our TrueRail technology.
Sphinx 101 audio plugin (macOS AU, Windows VST3), two visual skins (Amok, Nevy), default factory presets, this manual, signal test results, and your serial key.
macOS: Double-click the installer .pkg.
The plugin installs to /Library/Audio/Plug-Ins/Components/Sphinx 101.component (AU). The manual, signal test results, and an uninstaller .command are placed at /Applications/Analog Realism/Sphinx 101/ — visible in Finder, parallel to the Windows layout below. Your per-machine activation state (calibration.dat) lives in ~/Library/Application Support/Sphinx 101/ and is preserved across upgrades.
Windows: Run the installer .exe.
The plugin installs to C:\Program Files\Common Files\VST3\Sphinx 101.vst3 (VST3). The manual and signal test results are copied to C:\Program Files\Analog Realism\Sphinx 101\. Your per-machine activation state (calibration.dat) lives in %APPDATA%\Sphinx 101\ and is preserved across upgrades.
Open your DAW, insert Sphinx on any stereo bus.
Authorization (one-time): Click the key icon, paste your serial, click Authorize. A challenge code appears - copy it, visit the combined activation/deactivation web page shown, paste your serial and challenge on the Activate side, click Activate. Copy the response code back into Sphinx, click Activate. Done - zero internet needed from the plugin itself. Your serial works on 2 machines. Re-activating after an OS reinstall is seamless - the system recognizes your machine.
Deauthorization: Click the key icon, click Deauthorize. Copy the removal code, visit the same combined web page, switch to the Deactivate side, enter your email and removal code, copy the response back, and click Confirm removal from the plugin itself. Your seat is freed. Need the removal code again? Just repeat - the system re-issues it.
Demo Mode: Full functionality with a brief 2-second silence approximately every 1.5 minutes.
Requirements:
Insert Sphinx on your master bus, hit play. Try SLL (punch), Nevy (warmth), Amok (tube depth) main circuits. Push the Input TX Drive amount hard - that's real iron-core saturation you're hearing, modeled from actual transformer measurements. Switch transformer models and listen to how for example the bottom end changes character. Wiggle Width module to open the stereo image. Push the compressor threshold and feel the musical, natural glue. Try different EQ circuits to hear the true silky nature of real hardware EQs.
Most analog modeling plugins use "black box" modeling: measure what hardware does, build a filter that copies it. Sphinx models what the hardware IS - every transistor, tube, transformer core, capacitor, circuit and inductor simulated as individual components wired in the real circuit topology.
The difference is profound. In black-box models, the EQ doesn't know the compressor exists. In Sphinx, everything connects through a shared power rail. When the compressor works hard, it draws more current, causing voltage sag that shifts the bias of every other stage. The EQ's inductors saturate differently. The transformer's operating point shifts. The entire circuit breathes as one organism. This is TrueRail - twelve interacting mechanisms producing complexity that no hand-tuned algorithm can replicate.
All twelve are always active, all verified by automated measurement:
Audio passes through: Input Gain → Drive Stage → Input Transformer → Filters → Sub Bass Compressor → HF Compressor → HF SoftClip → Main Compressor → EQ → Width/Space → Output Transformer → Output Gain.
Input ─► [Input Gain] ─► [Drive Stage*] ─► [Input TX†] ─► [Filter] ─►
[Sub Comp] ─► [HF Comp] ─► [HF SoftClip] ─► [Compressor] ─►
[EQ] ─► [Width / Space] ─► [Output TX†] ─► [AutoGain] ─► [Output Gain] ─► Output
* Drive Stage = main circuit (SLL / Nevy / Amok) — always active
when the Input module is enabled.
† TX = Transformer (optional, independently bypassable per side).
┌──────────── TrueRail™ shared infrastructure (always active) ────────────┐
│ Power Supply Rail │ Crosstalk Matrix │ Thermal Drift │
│ Component Tolerance │ Instance Seed │ Summing-Node Bandwidth │
└─────────────────────────────────────────────────────────────────────────┘
Band-limited modules (Sub Comp, HF Comp, HF SoftClip) use sidechain-only processing - they detect energy in their frequency range and apply gain changes to the full-band signal but only in their own frequency areas, avoiding the phase artifacts of multiband splitting. Your audio is never chopped up and reassembled.
SLL - solid-state precision inspired by legendary British mixing consoles. Push-pull-style BJT topology modeled with the Ebers-Moll equations. The BJT's exponential transfer curve produces balanced even/odd harmonics (H2 ≈ H3). DriveStage THD ~0.08% at -6 dBFS. Measured rise time: fastest of all three circuits. Choose SLL when you want "finished" - polished and professional.
Nevy - single-ended Class-A discrete transistor with asymmetric off-centre bias (quadratic H2 enrichment) plus a gentle +0.4 dB low-shelf at 200 Hz modelling classic iron-core output-transformer coupling. The asymmetry lifts H2 about 5-8 dB above H3 — well above SLL's balanced harmonics while staying clearly below the tube path's depth. DriveStage THD ~0.26% at -6 dBFS. A true warm-transistor middle ground - not a tube clone. The compressor has a distinctive "hang" in its release - a slow initial decay that accelerates, creating the breathing quality real-life users and engineers love. Choose Nevy when you want "expensive."
Amok - all-tube signal path modeled with the Koren 12AX7 vacuum tube equations, full Miller capacitance, single-ended Class-A. Complex even-harmonic distortion, slowest rise time (Miller capacitance effect), deepest saturation. DriveStage THD ~3.67% at -6 dBFS, H2 dominates H3 by ~15 dB - dramatically richer 2nd-harmonic content than solid-state. The compressor uses tube variable-mu topology with soft knee (5-8 dB). Choose Amok when you want "alive."
Each circuit changes the entire chain - not just the drive stage, but the compressor character, EQ behavior, and transformer interaction.
The TrueRail tier scales the per-instance manufacturing-tolerance spread. Same instance seed across all tiers - switching tiers does NOT re-roll the random draw, it widens or narrows the same underlying variation pattern.
Tier A - Factory Spec. Tightest tolerances (±0.5-1% typical, scale factor 0.10). Closest match between L and R channels, smallest deviation from nominal component values. The "studio reference" sound - what a freshly-calibrated production unit would measure.
Tier B - Production Standard (default). Moderate tolerances (±1-2%, scale factor 0.15). Adds natural character without demanding attention. The sweet spot for most work.
Tier C - Vintage Wide. Widest tolerances (±2-4%, scale factor 0.20) - hand-wired vintage character. Most L/R asymmetry, most deviation from nominal values, the most "alive" sound. For material that benefits from organic, off-centre stereo imaging.
The buttons run left-to-right A → B → C, ascending from least to most variation - matching the "left=least, right=most" convention. Every plugin instance generates its own unique set of component tolerances - just like no two pieces of real hardware are identical.
Every value knob shares the same interaction set: drag vertically to adjust its value, hold Shift while dragging for fine adjustment, double-click a knob to type a value directly, scroll the mouse wheel for fine or fast adjustment depending on scroll speed, and Alt+Click (Option+Click on Mac) to restore the value the knob held before your last edit. The Alt+Click undo is single-step per knob — a second Alt+Click does nothing until you make another change. The undo state persists when you save and reload your DAW session.
Adjusts the input level into Sphinx with a range from -12 to +12 dB. Use Input Gain to hit the -6 dBFS sweet spot when necessary.
Hold Shift while dragging (or while scrolling the mouse wheel) to simultaneously counter-rotate the Output Gain by the same amount in the opposite direction — drive the plugin harder without changing the output level. Useful for auditioning saturation and colour without volume bias.
Four physically distinct transformer models, available on both input and output positions. Each is modeled from real core materials with different magnetic properties:
M1166 - silicon steel core. Warm, smooth saturation. Resonant peak around 21 Hz adds subtle low-end weight. The classic "console sound." Best for: warmth and body.
C9049 - nickel alloy core. Authoritative, controlled saturation. Resonant peak around 35 Hz gives a tighter, more defined low end. Best for: precision and punch.
K1166 - grain-oriented silicon steel. Tight, fast saturation with minimal overshoot. Best for: transient-heavy material that needs transformer color without smearing.
L1544 - amorphous core (that Swedish one). The most transparent. Symmetric B-H loop produces odd-harmonic-dominant character (H3 greater than H2) - physically accurate to real amorphous core behavior. Best for: transparency with a hint of iron.
The input transformer uses full Jiles-Atherton magnetic hysteresis - it remembers its recent magnetization history, producing saturation that depends on what came before. The output transformer uses a matched harmonic profile without the memory effect, preventing cross-frequency modulation and comb-filtering between stages.
| Control | Range | What It Does |
|---|---|---|
| Drive | 0% to 200% | How hard you push the transformer core. 100% = calibrated match to real hardware. Below 100% = cleaner than hardware. Above 150% = audible saturation. |
High-pass and low-pass filters with circuit-modeled rolloff. These model the actual component behavior of passive filter networks. Positioned early in the signal chain - before the compressors and EQ - so they clean up the frequency extremes before dynamics processing acts on them. Like the EQ, these aren't just digital biquads - each circuit topology models the actual component behavior of real filter networks, including the way source and load impedances interact with the filter elements.
Pultey (Passive LC): - the gentlest slope. Modeled after passive inductor-capacitor filter networks with their naturally smooth rolloff and subtle phase character. The HPF has a gentle resonant bump near the cutoff that adds a slight "lift" just above the filter point - cleaning up the sub without thinning the low end. Best for: mastering, where transparency matters more than precision.
Nevy (Discrete Class-A): - steeper rolloff with the warm, slightly rounded character of discrete transistor filter circuits. The real component tolerances create a filter that isn't mathematically perfect - and that imperfection is what makes it sound musical rather than clinical. Best for: general mixing, cleaning up recordings without sterile artifacts.
Sllamok (Active Op-Amp): - hybrid of SLL/Amok circuits. Clean Butterworth-derived response with predictable, measured behavior. Minimal phase disturbance in the passband. Best for: surgical cleanup, post-production, situations where the filter must be invisible.
Maney (Tube Buffer): - passive filter network followed by a tube buffer stage. The tube adds subtle 2nd-harmonic warmth to whatever passes through the filter, so even the act of filtering becomes a tonal enhancement. Best for: creative filtering, when you want the HPF or LPF to add character rather than just subtract frequencies.
Sharp mode: Activates a steeper rolloff slope with a tighter Q at the cutoff frequency. In normal mode, the filters roll off gently with broad, musical curves. In Sharp mode, they cut more aggressively - useful for removing problematic low-end rumble or taming harsh high-frequency content without affecting the nearby frequencies. The Sharp HPF shows a measured peak of +1.4 dB at the cutoff frequency - a subtle resonant lift that prevents the bass from sounding "scooped" even with aggressive filtering.
| Control | Range | What It Does |
|---|---|---|
| HighCut | 16 kHz to 22 kHz | Everything above this rolls off. Tame harshness and air. |
| LowCut | 15 to 200 Hz | Everything below this rolls off. Clean up rumble and mud. |
| Mode | Default / Sharp | Steepens the filter slope for more aggressive filtering. |
| Mode | PreTX OFF/ON | Routes the filter after/before the input transformer. |
Each compressor module in Sphinx 101 includes their own selection of four circuit topologies, each modeled from real hardware behavior:
SLL (VCA) - fast, punchy, predictable. Hard knee (1-3 dB). Single exponential release. The "glue" compressor. Best for: pop, rock, electronic - anything that needs rhythmic punch.
Nevy (Transformer-coupled) - warm, musical, forgiving. Medium knee (3-5 dB). Dual time constant release with characteristic "hang." Best for: jazz, soul, acoustic - organic material that needs cohesion without aggression.
Amok (Tube) - thick, deep, saturated. Soft knee (5-8 dB). Tube-dependent harmonics increase with gain reduction depth - the harder it works, the more harmonics it adds. Best for: dense mixes, hip-hop, cinematic - when you want the compressor to be heard.
Maney (Vari-Mu) - gentle, program-dependent, self-adjusting. Softest knee. Hyperbolic release (slower than exponential). The tube gain element shifts operating point with signal level, making it naturally program-dependent. Best for: mastering, classical, delicate material.
Controls the low-end foundation. Operates via sidechain - detects energy below the crossover frequency and applies gain changes only to its own frequency area of the full signal, so your low end gets tighter without multiband phase artifacts.
| Control | Range | What It Does |
|---|---|---|
| Threshold | -18 to +6 dB | Where sub compression begins. |
| Crossover | 40 to 250 Hz | Frequencies below this are controlled. Higher = wider bass management. |
| Attack | 1 to 500 ms | Catch bass transients before they eat headroom. |
| Release | 1 to 1000 ms | Settle between kick hits at typical tempos. |
| Makeup | -12 to +12 dB | Compensate for gain reduction in the affected frequency-area only. 12 o'clock = 0 dB (unity). |
| Parallel | 0% to 100% | Blend. Lower values keep the natural bass weight while adding control. |
| Mode | Soft / Hard | Soft knee for gentle compression, hard knee for aggressive character. |
| Mode | PostComp OFF/ON | Routes the module before/after the compressor module. |
Tames high-frequency energy - sibilance, harsh cymbals, brittle high-mids. Sidechain detects energy above the crossover frequency and applies gain changes only to its own frequency area of the full signal. Your highs get smoother without the artifacts of traditional de-essing.
| Control | Range | What It Does |
|---|---|---|
| Threshold | -18 to +6 dB | Where HF compression begins. |
| Attack | 1 to 500 ms | Fast enough for sibilant peaks. |
| Release | 1 to 1000 ms | How recovers before the next HF transient, keeping the top end alive. |
| Crossover | 6000 to 20000 Hz | Frequencies above this are controlled. |
| Makeup | -12 to +12 dB | Compensate for gain reduction in the affected frequency-area only. 12 o'clock = 0 dB (unity). |
| Parallel | 0% to 100% | Blend for natural HF taming. |
| Mode | Soft / Hard | Soft knee for gentle compression, hard knee for aggressive character. |
| Mode | PostComp OFF/ON | Routes the module before/after the compressor module. |
A saturation-based high-frequency limiter - think of it as the analog equivalent of a tape machine's natural HF compression. Two circuits:
Tube - warm, rounded saturation. Even harmonics. Smooths harsh peaks into musical warmth. Sidechain detects energy above the crossover frequency and applies gain changes only to its own frequency area of the full signal.
DPX - tighter, more aggressive clipping inspired by the Distressor's British mode. Adds edge and presence while preventing harshness.
Timing is fixed - this is a clipper circuit, not a compressor. The response is inherent to the saturation characteristic, producing instant, transparent peak control.
| Control | Range | What It Does |
|---|---|---|
| Threshold | -18 to +6 dB | Where soft clipping begins. Lower = more saturation. |
| Crossover | 6000 to 22000 Hz | Frequencies above this are controlled. |
| Makeup | -12 to +12 dB | Compensate for gain reduction in the affected frequency-area only. 12 o'clock = 0 dB (unity). |
| Parallel | 0% to 100% | Blend. Even small amounts add polish to the top end. |
| Mode | PostComp OFF/ON | Routes the module before/after the compressor module. |
| Control | Range | What It Does |
|---|---|---|
| Threshold | -18 to +6 dB | Where compression begins. Lower = more compression. |
| Crossover | 20 to 400 Hz | For listener circuit only - still compresses the full-band. |
| Attack | 1 to 500 ms | How fast the compressor reacts. Faster = catches transients. Slower = lets punch through. |
| Release | 1 to 1000 ms | How fast the compressor lets go. Faster = more aggressive. Slower = smoother, more musical. |
| Makeup | -12 to +12 dB | Compensate for gain reduction. 12 o'clock = 0 dB (unity). |
| Parallel | 0% to 100% | Blend with unprocessed signal. Lower values preserve transients while adding density. |
| Mode | Soft / Hard | Soft knee for gentle compression, hard knee for aggressive character. |
Sphinx 101's three-band EQ is not just another digital filter. Each band contains a real inductor model with saturation characteristics - at moderate settings it's a clean, precise EQ, but push the gains harder and the inductors begin to saturate, adding harmonic complexity that's physically generated, not artificially added. Four circuit topologies offer fundamentally different approaches to equalization:
Sllamok (Active Op-Amp): - hybrid of SLL/Amok circuits, clean, precise, zero coloration. Constant-Q design: the bandwidth stays the same regardless of how much you boost or cut. No inductor saturation - this is a purely mathematical EQ implemented with modeled op-amp circuits. THD below 0.01% at any setting. The surgical option. Best for: corrective work, precise notching, clinical frequency adjustments where you want zero tonal footprint from the EQ itself.
Nevy (Discrete Class-A with Inductors): - warm and musical with signal-dependent character. Proportional-Q: small boosts are broad and gentle, large boosts narrow and focused - the way vintage British console EQs naturally behave. Real inductor saturation adds warmth at high boost levels (THD rises from 0.01% to 0.5% as you push harder). Swinging inductance shifts the center frequency by 1.5% with signal level - the subtle modulation effect that's the recognizable signature of vintage inductor EQs. Best for: musical tone shaping, adding warmth to vocals, giving guitars presence - anything that benefits from the EQ having its own character.
Pultey (Passive LC + Tube Buffer): - the strongest inductor saturation of all four circuits, modeled after the revered American passive equalizer. H2/H3 harmonic ratio of 23:1 - overwhelmingly rich 2nd-harmonic warmth. 4% swinging inductance. The famous passive-EQ "boost and cut at the same frequency" trick works here: simultaneously boosting and cutting the same band creates a scooped-midrange-with-boosted-sub curve that's impossible with conventional EQ. The inductors saturate at the right point and in the right way - by design, acting as a cross between an EQ and a low-frequency limiter. In A/B comparison with a leading commercial emulation, Sphinx's Pultey matched within 0.7 dB THD and 1.3 dB H2 level. Best for: mastering sweetening, adding weight and air simultaneously, the classic tube passive EQ sound that has defined hit records for sixty years.
Maney (Passive Parallel Topology): - modeled after a celebrated American tube passive mastering equalizer with a unique parallel band architecture. Unlike conventional series EQs where each band stacks on top of the previous one, the parallel topology sums bands independently - boosting four bands at the same frequency caps at around +20 dB total, not +80 dB. This "soft addition" means you can make aggressive-looking settings without the output becoming ugly. Strong inductor saturation with musical character. Best for: mastering, broad tonal sculpting, situations where you want to use multiple overlapping bands aggressively without the sound falling apart.
Three-band EQ with real inductor saturation modeling. At moderate settings, it's a clean, musical EQ. Push the gains harder and the inductors begin to saturate, adding harmonic complexity that static EQ plugins can't produce - because in Sphinx, the EQ components are actually saturating, not just filtering.
Four circuit topologies match the main circuit selection, each with subtly different Q behavior and saturation character.
| Control | Range | What It Does |
|---|---|---|
| Low Gain | -12 to +12 dB | Boost/cut the low shelf. 12 o'clock = flat. |
| Mid Gain | -12 to +12 dB | Boost/cut the mid band. |
| Hi Gain | -12 to +12 dB | Boost/cut the high shelf. |
| Low Freq | 20 Hz to 400 Hz | Low shelf corner frequency. |
| Mid Freq | 200 Hz to 12 kHz | Mid band center frequency. |
| Hi Freq | 4 kHz to 18 kHz | High shelf corner frequency. |
| Mode | Default / Sharp | Softer curve for gentle corrections, or harder curve for more aggressive, surgical character. |
| Mode | PreComp OFF/ON | Routes the module after/before the compressor module. |
Three controls for psychoacoustic spatial enhancement, inspired by the dimensional processing found in classic hardware spatial processors.
Width (range: -100% to +100%) — tube-stabilized stereo expander. Positive values widen the stereo image by processing the side signal through a 5751 triode stage that adds subtle 2nd-harmonic warmth to off-center content, psychoacoustically stabilizing the center image as the sides expand. Negative values narrow toward mono while retaining the tube's warm character — like pulling down the stereo bus width on a real console where iron and tubes are still in the signal path. At 0% the tube is bypassed: fully transparent. The sweet spot for most masters is +10% to +40%. The tube participates in the TrueRail shared power rail — when the compressor pumps hard, Width's spatial enhancement breathes with the dynamics. 12 o'clock = 0% (no change).
Space (range: -100% to +100%) — frequency-targeted spatial enhancement. Opens or focuses the stereo image by gently shaping the side signal's presence range (a broad bell centered around 3 kHz, the 2–5 kHz band where human spatial perception is most acute). Positive values boost that band on the sides — off-center elements like panned guitars, background vocals, and room ambience feel wider and more open. Negative values cut it, pulling the image toward a more focused, compact center. The shaped side then feeds the Width tube, gaining analog harmonic character. It is a steady-state filter — no modulation, no artifacts. At 0% fully transparent. Effective at subtle settings (+5% to +30%). 12 o'clock = 0% (no change).
Mono Sum (range: 20 to 200 Hz) — frequencies below the set value are smoothly summed to mono. Essential for vinyl cutting, club systems, and tightening the sub-bass foundation. The filter is transparent and artifact-free. Default: 75 Hz.
Use Output Gain for final level matching back into DAW. Ranges from -12 to +12 dB.
Hold Shift while dragging (or while scrolling the mouse wheel) to simultaneously counter-rotate the Input Gain by the same amount in the opposite direction.
Enabled by default. Unlike conventional auto-gain systems that measure input vs. output levels (which fights compression and misjudges silence), Sphinx 101's AutoGain operates per-module — each processing stage independently estimates and compensates its own gain contribution, preserving the natural interaction between modules.
It uses parameter-based prediction for frequency-shaping modules and a slow (~6-second) running-average of actual gain reduction for dynamics modules. Crucially, AutoGain restores the average level only — the per-sample envelope is left untouched, so rhythmic pumping, transient character, crest factor, and dynamic shape are all fully preserved.
Not a limiter, not a signal follower — a smart level anchor. Turn off when targeting a specific output loudness for delivery. Does not compensate the Output Gain knob — that remains your final manual level control. Manual makeup gain knobs in each module add on top of AutoGain's correction.
The master effect intensity control. This does NOT digitally mix wet and dry signals, which would cause massive comb filtering with as realistic and chaos-like signal results as Sphinx produces for its signature analog sound. Instead, it scales each module's processing intensity: at 0%, all modules operate at their default/bypass values. At 100%, all modules operate at their user-set values. At 50%, every parameter sits halfway between its default and your setting — a true intensity scale, not a volume blend. One signal path at all times, zero phase cancellation. This is why Sphinx's parallel mix sounds clean at every position. The parallel mix knobs are more like "Strength" knobs, which tone up or down the effect of the module or the whole plugin itself.
Oversampling processes the strongest nonlinear stages at a higher internal rate to reduce aliasing. The DriveStage (main BJT/tube saturator, FIR equiripple halfband) and both transformers (Input TX and Output TX, IIR halfband) run at the user-selected oversampling rate (2×/4×/8×/16×). The remaining gentler nonlinearities — EQ, Compressor color, and Width — operate inline at base sample rate; their distortion (~0.004% THD) produces aliasing at −120 to −140 dB, far below audibility, so spending CPU oversampling them buys nothing measurable. All four levels are fully functional and user-selectable. Higher rates give cleaner antialiasing on the oversampled stages at the cost of CPU. The default 2× is excellent for most work. Use 4× or higher for final masters, especially when pushing the saturation stages hard. At the default 2× the actual aliasing in the oversampled stages is suppressed to about −120 dB (measured in isolation), far below TrueRail's ~−70 dB analog noise floor. The measured full-chain floor is therefore identical across all oversampling factors because that −70 dB analog noise (thermal drift, rail sag) masks the aliasing completely. Higher oversampling factors DO suppress aliasing further — the suppression simply becomes invisible below the analog character floor. This is the same behavior as measuring aliasing on real hardware: the console's own noise floor is always the limiting factor, not the digital conversion.
| Setting | Visible Floor* | Actual Aliasing | CPU Impact | Best For |
|---|---|---|---|---|
| 2× (default) | −74 dB | −120 dB | Low | Most work |
| 4× | −74 dB | < −120 dB | Moderate | Final masters, critical listening |
| 8× | −74 dB | < −120 dB | High | Heavy saturation with HF content |
| 16× | −74 dB | < −130 dB | Very high | Offline rendering of final masters |
\* The visible floor is the modeled analog noise (TrueRail power supply + ThermalDrift), not aliasing. Actual aliasing sits ~50 dB below this floor at the default 2× — invisible because the analog noise modeling is louder, exactly as on real hardware.
Calibrated to real hardware ballistics - 300 ms rise time, 1.25% overshoot (VU mode); 1 ms attack, 2.8 s decay (PPM mode). LED indicators use smooth opacity breathing, never binary on/off. VU meters calibrated to -6 dBFS = 0 VU (the recommended operating level). Gain reduction meter for real-time compression depth. Spatial LED phase indicator. Knob value display on hover and adjust.
Switch instantly between skins via the skin selector "handle area" in the plugin's GUI faceplate.
Amok (default) - light-styled metal with industrial accents.
Nevy - Alternative aesthetic, same controls, dark-styled layout.
Sphinx operates at -6 dBFS peak. This is where the nonlinear components produce their most musical harmonics.
Too hot (peaking at 0 dBFS): harsh highs, pumping, lost definition - reduce Input Gain 3-6 dB.
Too cold (below -12 dBFS): thin and lifeless - increase Input Gain.
The TX Drive knobs let you push transformer saturation independently: 100% is the calibrated match to real hardware measurements, past 150% for obvious warmth, below 100% for cleaner-than-hardware operation.
Sphinx 101 models analog circuits at the individual component level. Each electronic part is computed from published physics equations - not generic saturation curves or static waveshapers. The componenttest suite verifies every model directly against its expected behavior; the sections below name the model and list what is measured.
The Amok circuit's triode stage uses Norman Koren's plate current equations, derived from real 12AX7 tube characteristics. The model computes plate current as a function of grid voltage, plate voltage, and tube-specific parameters (mu, kp, kvb, Ex). This produces the asymmetric transfer curve that gives tubes their characteristic even-harmonic warmth - H2 rises naturally with signal level, exactly as in real hardware. Miller capacitance is modeled for accurate HF rolloff, and a cathode-bypass capacitor sets the operating bias.
Verified: plate current accuracy, amplification factor (mu), H2 enrichment vs level, THD-rises-with-level curve, output level still alive at hot drive, per-instance seed variation in tube parameters.
The SLL and Nevy circuits use BJT gain stages based on the Ebers-Moll equations. The model computes collector current from base-emitter voltage using the transistor's forward current gain (hFE), Early voltage, and thermal voltage. The SLL topology uses push-pull configuration (symmetric, odd-harmonic dominant). The Nevy topology adds a DC bias offset for Class-A operation, producing the asymmetric transfer curve that enriches even harmonics. Two real part numbers are modeled — BC184 (NPN, low-noise audio) and BC214 (PNP complement) — with the Ebers-Moll equations populated from their datasheets.
Verified: BC184 THD at -6 dBFS, BC214 THD at -6 dBFS, NPN vs PNP transfer-curve differential (they should not be identical), monotonic THD vs level, output level alive under drive, per-instance seed variation.
The SLL compressor uses a voltage-controlled amplifier based on the Blackmer/THAT design. The model computes gain as an exponential function of the control voltage, providing dB-linear gain control. THD at unity gain and gain accuracy at attenuated settings are modeled from published specifications.
Verified: unity-gain output level (calibrated to -12 dBFS test point), -10 dB attenuation point sits at the expected -22 dBFS, monotonic THD ordering across grade tiers, low-level output stays above the noise floor.
Active EQ stages use an op-amp model based on the NE5534 - the industry standard for analog audio. The model includes finite gain-bandwidth product (10 MHz), open-loop gain (100 dB), output slewing into the supply rails, and the resulting THD from bandwidth-limited negative feedback at high frequencies.
Verified: unity-gain transfer at 1 kHz tracks the input voltage, finite-GBW HF rolloff visible above audio band, output clamps at the supply rails (slew/clip ceiling), per-instance seed variation in component values around the op-amp.
Resistors, capacitors, and inductors are modeled with per-instance tolerance variation. Each plugin instance receives a unique seed that determines its component values within the TrueRail tier's tolerance range:
Metal-film resistors show tighter tolerance than carbon-composition; film capacitors hold value more closely than electrolytics. Inductors exhibit swinging behavior — inductance decreases with signal level via the Jiles-Atherton current-dependent saturation curve, which affects EQ response under high-level transients. Johnson (thermal) noise scales with resistance, as in real hardware.
Verified: metal-film 1% resistor standard deviation falls inside the tolerance band, carbon-composition spread exceeds metal-film, film capacitor 1% spread inside tolerance, electrolytic spread exceeds film, inductor saturates monotonically with current, Johnson noise scales with R.
Four transformer core materials are modeled using the Jiles-Atherton magnetic hysteresis equations (see also section 9 — Transformers). The JA model computes magnetization from field intensity using five physical parameters: saturation magnetization (Ms), anhysteretic shape (a), domain coupling (alpha), pinning/loss coefficient (k), and reversibility (c). Each core material has different JA parameters producing different B-H loop shapes — M1166 silicon steel is asymmetric and H2-rich; L1544 amorphous is symmetric and H2-suppressed.
Verified: M1166 H2/H3 ratio asymmetric (>1.0), M1166 more H2-dominant than L1544, measurable M-vs-L spread in the H2/H3 ratio, per-instance JA-parameter tolerance variation non-zero.
VU, PPM, and LED meters use standard-compliant ballistics:
Verified (VU): needle rests at zero with no input, 0 VU lands inside [0.7, 1.3] for the -6 dBFS reference tone, needle rises measurably over 300 ms. (PPM): needle at zero on silence, attack catches a transient within 20 ms, sustained tone holds high. (LED): opacity responds to level, attack is smooth (not stepped), sustained signal settles high.
A 30-probe runtime audit (RV01-RV30) drives the full plugin through pathological inputs — zero-length buffers, single-sample buffers, transport jumps, repeated prepareToPlay calls, full-scale and hot signals, every knob at min/max, rapid circuit-switch crests, skin reloads, preset restores, two simultaneous instances — and asserts that every output sample is finite, bounded, and click-free. The same audit verifies AutoGain doesn't kill level, flat-EQ doesn't color, parallel mix doesn't comb, mono-sum stays smooth, the sidechain HPF does not leak into the audio path, and the entire plugin produces bit-identical output run-to-run (deterministic seeding).
Verified: 30 robustness probes pass on every build (see the Signal Verification chapter for the full list).
Beyond the per-component models, six "chassis" subsystems are always on - they are how the components interact, not what they do individually:
Formats: Audio Unit (AU) for macOS, VST3 for Windows.
Platform: macOS 11 (Big Sur) or later — separate Apple Silicon (arm64) and Intel (x86_64) installers; the Intel build also runs on Apple Silicon via Rosetta 2. Windows 10 or later (x64).
Sample rates: 44.1-192 kHz (verified identical character).
Buffer sizes: 64-2048 (verified bit-identical output).
Internal precision: 64-bit.
Oversampling: 2x/4x/8x/16x.
Aliasing suppression at 2×: −120 dB (50 dB below the analog noise floor)
Noise floor: -110 dBFS.
Operating level: -6 dBFS.
Gain structure: unity loudness at default settings — output RMS tracks input within about ±0.5 dB at the -6 dBFS sweet spot. (Peak crest expands a few dB through the chain from cascaded analog phase rotation and oversampling inter-sample reconstruction, exactly as a real console strip does — this is loudness-transparent, not a gain error, and is verified on real program material.)
Bypass transparency: -68.28 dB residue (TrueRail chassis character).
Polarity: non-inverting at all frequencies.
Latency: sample-accurate DAW compensation.
Parameters: 72, all automation-compatible.
TrueRail mechanisms: 12 simultaneous.
Authorization: one-time web visit, serial + C/R, 2 seats per serial.
Demo mode: 2 sec silence in approximately 1.5 minute intervals.
Every copy is built from the same codebase that passed 2,043 automated signal measurements across ten categories: full signal chain integrity (977 measurements - includes the AG01 - AG07 envelope-preservation tests and PhaseIntegrity lock-in tests), compressor dynamic behavior (197), EQ accuracy (157), transformer verification (217 — four iron cores: M1166 / C9049 / K1166 / L1544), module isolation (130), parameter sweep verification (168), oversampling and anti-aliasing (60), component-model verification with 30 edge-case robustness tests (68), configuration audit (63), and music-material integrity (6, fulltest suite). Zero failures, zero warnings across every suite. Test results are included with your installation.
Beyond the synthetic test signals (sine, multitone, noise), every suite now also runs a real-music level section: a calibrated commercial master is pushed through the actual welcome preset and metered the way a DAW meter does. This is the authoritative loudness check — mono and synthetic tones can mislead on level (a mono tone collapses the stereo side and hides the chain's true behaviour), so the design's unity-loudness invariant is confirmed against real, bass-heavy, stereo program material alongside the synthetic measurements.
Sphinx is built to be measured. If you open it in an analyzer such as Plugin Doctor and inspect its harmonic distortion, Hammerstein kernels, or phase response, this section explains exactly what you are seeing and why — with the detail of a hardware technical sheet. None of the behaviours below are flaws; each is the physical consequence of modeling real console circuitry at the component level.
The three console personalities model three classes of legendary large-format console hardware. We use original names — SLL, Nevy, Amok — out of respect for the trademark holders, but the topologies and the measured harmonic signatures below identify the sonic lineage to any engineer who knows the genre:
All values below are taken at the −6 dBFS operating sweet spot with the default preset, the same conditions an analyzer uses by default.
The three personalities produce distinctly different, deliberate THD — this is musical harmonic content from the modeled circuitry, not digital distortion:
| Personality | THD (≈) | In percent | Character |
|---|---|---|---|
| SLL | −62.0 dB | 0.08 % | Balanced even/odd harmonics — the lowest distortion of the three; clean, forward, defined. |
| Nevy | −51.7 dB | 0.26 % | 2nd-harmonic-dominant — the body and warmth of transformer-coupled gain. |
| Amok | −28.7 dB | 3.67 % | Rich, predominantly 2nd-harmonic — the lush, saturated valve character. |
The harmonic balance is also visible in the Hammerstein-kernel G2/G3 levels, which rise with the "warmth" of the topology: SLL G2 ≈ G3 (balanced — the BJT's exponential transfer curve), Nevy G2 above G3 by ~5–8 dB (transformer-coupled H2 enrichment), and Amok G2 above G3 by ~15 dB (the 12AX7 triode's asymmetric transfer curve). These values match the harmonic profiles of the real console classes they model.
The noise floor. Below the harmonic spikes sits a smooth floor from the TrueRail shared power-supply model and the ThermalDrift component-aging LFOs — the modeled equivalent of a real analog supply and thermally-varying parts. It measures roughly −70 to −74 dB and is well below audibility. Crucially, this floor is identical at every oversampling rate (2× through 16×) — which proves it is analog modeling, not digital aliasing. Real aliasing would fall as oversampling rises; analog noise does not.
What to look for. Clean harmonic spikes at exact integer multiples of the test frequency, rising above a smooth floor. Any non-harmonic spikes or irregular comb patterns would indicate aliasing — Sphinx shows none in the audio band.
A Hammerstein decomposition separates the nonlinearity into frequency-dependent polynomial kernels — G1 is the linear (fundamental) response, G2 the 2nd-order term, G3 the 3rd-order, and so on. Each personality has a distinct fingerprint:
Low-frequency behaviour. Below ~50–200 Hz the higher-order kernels roll off, and on the Amok personality they show a characteristic oscillation. This is the interaction of the coupling capacitors with the valve stage's plate-decoupling and cathode-bypass networks at sub-audio frequencies — the same behaviour measurable in real tube equipment with iron coupling, and absent from the solid-state SLL and transformer-coupled Nevy paths.
High-frequency behaviour. Above ~5–10 kHz the kernels converge into the noise floor. This is physically correct: real transformers and amplifiers are bandwidth-limited, which naturally suppresses high-frequency harmonic generation.
Sphinx's phase response is dominated by its transformer models. The DriveStage (main circuit) contributes effectively zero phase rotation in the audio band — its oversampling uses a linear-phase FIR halfband filter that adds only pure delay, fully compensated by the DAW's plugin delay compensation. Each transformer (Input TX and Output TX) contributes approximately one full rotation ("wrap") from its reactive elements: the coupling capacitor, winding inductance and resistance, and shunt capacitance that form the transformer's linear shell model. With both transformers enabled, the full chain shows roughly two wraps — monotonic downward rotation with no direction reversals.
The band-split compressors (Sub Comp, HF Comp, HF Softclip) are deliberately sidechain-only — their crossover filters act on a detection copy, never the main audio, so they add no phase rotation to the output. The EQ, Compressor color saturation, and Width modules operate inline at base sample rate without oversampling, contributing no additional phase. The measured phase is almost entirely the transformers.
At the highest frequencies (above ~18 kHz), a small amount of additional phase rotation appears from the IIR halfband anti-aliasing filters on the Input and Output Transformer oversamplers. This is inaudible and confined to the extreme top of the spectrum.
Because the phase rotation comes primarily from the transformers rather than the DriveStage, the three main circuits (SLL, Nevy, Amok) produce very similar phase responses. Subtle personality-specific differences appear at low frequencies from each circuit's bias point and gain structure interacting slightly differently with the transformer's reactive load.
Phase rotation and peak levels. Even with modest phase rotation, the frequency-dependent phase shifts from the transformer models reshape the waveform of complex program material. On music this can raise instantaneous peak levels by roughly 3–5 dB while RMS (loudness) stays at unity — exactly what happens when you run audio through a real analog console with iron transformers. It is not a gain error: a DAW peak meter will read higher, but a loudness (LUFS) meter will read the same as the input. AutoGain holds RMS unity; the peak rise is the physical consequence of frequency-dependent phase rotation on a complex waveform.
The strongest nonlinear stages — the DriveStage (main BJT/tube saturator) and both transformers (Input TX and Output TX) — run at the user-selected oversampling rate (2× / 4× / 8× / 16×). The DriveStage uses an FIR equiripple halfband for steep, linear-phase HF stopband control on its sharp wave-shaper; the transformers use IIR halfband filters. The remaining nonlinearities — EQ, Compressor color, and Width — operate inline at base sample rate. Their gentle distortion (~0.004% THD) places aliasing at roughly −120 to −140 dB, far below the analog noise floor and below audibility, so oversampling them would burn CPU for no measurable benefit.
At the default 2× oversampling, aliasing products are suppressed to approximately −120 dB — well below the plugin's analog noise floor of about −70 dB. Measured in isolation (a saturating stage with the analog noise sources removed) the foldback drops from roughly −38 dB un-oversampled to about −120 dB at 2×, and stays in that −120 to −130 dB range at 4× / 8× / 16×. The visible floor in a full-chain measurement (~−70 to −74 dB) is therefore entirely the modeled analog noise — the TrueRail power supply and ThermalDrift component-aging — not digital aliasing. Higher oversampling rates push aliasing even further below this floor; the improvement is simply no longer visible, because the modeled supply and thermal noise are the limiting factor (exactly as on real hardware). The flatness of that floor across all four oversampling rates is itself the proof that what you measure is analog character, not digital aliasing.
The oversampling filters are phase-transparent in the audio band: the excess phase contributed by the oversampling system measures 0.0000 rad across the midband (verified with dedicated phase-audit tooling). The oversampling adds no digital phase fingerprint on top of the analog modeling — the phase behaviour you measure is entirely the modeled circuit.
Does the plugin have a guide inside the DAW? Yes, click on the book icon on the lower-right corner of Sphinx GUI to open a popup guide.
Transfer license? - Deauthorize old machine, activate new.
Can't access old machine? - Email support.
Lost response code? - Repeat the activation or deactivation steps - it's re-issued.
Internet required? - One-time web activation only, from any computer. Plugin does not call home, ever, because everyone deserves privacy.
Server down? - Existing installs unaffected.
Mac + Windows? - Both ship today (AU for macOS, VST3 for Windows). Each install counts as one seat.
96/192 kHz? - Verified identical character.
CPU? - Around the usual industry topdogs.
Multiple instances? - Yes, each gets unique component tolerances. Multiple instances never null due to their analog, unpredictable nature.
Plugin not found: - rescan plugins.
Clicks on DAW transport location changes: - increase buffer.
High CPU: - try 2x oversampling.
Sounds congested: reduce input 1-3 dB.
Challenge/Response rejected: - copy exactly, it's case-sensitive. Or email support with your serial and email.
Max seats reached: - deauthorize old machine or email support.
Air-gapped studio: - you can use any internet-connected computer with the web activation page to get the codes.
Sphinx 101 developed by Analog Realism. Component models based on published scientific equations: Koren (1996), Jiles-Atherton (1986), Ebers-Moll (1954). Made in Finland. (C) 2026 Analog Realism. All rights reserved.