SCMThe best master is yet to come…
Sound Chain Master — Professional Audio DSP · Component Lab — is a focused mastering and signal-chain toolkit. This booklet collects, in one printable reference, the educational material behind every panel of the application: the history, purpose, key parameters, transfer equations, mnemonic diagrams and signal-chain placement of each processor.
A signal chain is the ordered sequence of processors a sound passes through — corrective equalisation first, then dynamic control, then colour and space, and finally level. The order matters as much as the processors themselves: the same EQ and compressor sound different in different positions, and effects such as reverb and delay are usually better in parallel so the dry signal stays untouched.
Every entry follows the same structure so processors can be compared on equal terms. History traces where each technique came from. Why it’s used explains its job. Key parameters maps the controls to the sound. The Transfer equation gives the maths where it is useful. The Diagram visualises the behaviour. And Where it belongs fixes the processor in the chain.
Loudness is treated throughout with the modern, perceptual standards. LUFS (ITU-R BS.1770) measures perceived loudness with a K-weighting filter across Momentary, Short-Term and Integrated windows. The Bob Katz K-System anchors the meter’s 0 dB mark at a headroom reference below 0 dBFS — K-12 for broadcast, K-14 for pop and home mastering, K-20 for the most dynamic material — so well-calibrated programme material sits around 0 dB and the headroom above it is visible at a glance. Streaming platforms normalise to roughly −14 LUFS; the days of chasing 0 dBFS peaks are over.
Use this booklet as a learning resource and a quick reference. Read the panels in chain order to see how a master is built; jump to a single processor to recall a control or a formula; and keep the references at the back for deeper study.
Equalization is as old as the telephone — early 20th-century engineers at Western Electric and Bell Labs built “line equalizers” to compensate for the high-frequency loss of long cables.
Studio “tone controls” appeared on 1930s cinema and broadcast consoles. The parametric EQ — where every band lets you set frequency, gain AND bandwidth (Q) independently — was invented by George Massenburg in 1969 and commercialized in the ITIParametric.
An EQ changes the balance of frequencies in a signal — cutting mud, lifting clarity, removing rumble or shaping tone.
It is the single most important corrective and creative tool in mixing.
| Band | Range | Boost → | Cut → |
|---|---|---|---|
| Sub-bass | 20–60 Hz | Power & felt weight — the chest-thump of a kick or bass. | Cleans rumble, mic handling noise and DC-like mud; protects headroom. |
| Bass | 60–250 Hz | Warmth, fullness and fundament of bass instruments. | Tightens a flabby low end and removes boom from a kick or bass. |
| Low mids | 250–500 Hz | Body, wood and "cardboard" thickness. | Clears muddiness and boxiness — the most common corrective cut. |
| Mids | 500 Hz–2 kHz | Presence, honk and nasal forwardness. | Removes nasal/honk and de-clutters a cluttered mix. |
| High mids | 2–4 kHz | Attack, definition and the "edge" that helps things cut. | Softens harshness and takes the fatigue off bright sources. |
| Presence | 4–6 kHz | Clarity and bite, especially on vocals and snare. | Reduces harshness and listener fatigue on bright material. |
| Sibilance | 6–8 kHz | Air and breath on vocals; brightness on cymbals. | De-esses vocals and tames sharp cymbal harshness. |
| Brilliance | 8–12 kHz | Sheen, openness and perceived detail. | Hides tape/tape-style hiss or dulls a too-bright top end. |
| Air | 12–20 kHz | Sparkle, space and "expensive" openness. | Tames digital harshness or brittle highs. |
EQ belongs near the start of every chain and on every bus.
Dynamic EQ merges two ideas that matured separately: the parametric EQ (Massenburg, 1969) and the frequency-conscious compressor or “de-esser” (1980s, e.g. the Orban 526 / 622A).
Engineers needed a tool that only cuts a frequency WHEN it gets too loud, not all the time — to tame sibilance, a snare ring or a resonant bass note without deadening the sound at rest.
Dynamic EQ is used for problems that are only problems some of the time: a vocal that gets harsh on loud words, a bass that booms on certain notes, a snare with an occasional ring.
Unlike static EQ it never removes the frequency when it is quiet, so it preserves the natural tone.
Each band wraps a tiny saturator around its compressor (a de-esser: out = dry + comp(band) − compIdle(band)). The per-band saturator is deliberately far gentler than the main Saturation panel: there is no pre-boost, and the curve drive coefficient is 0.2× — so a band can warm its zone without distorting it. Because there is no pre-boost, the plotted transfer curve is exactly what you hear.
none — satDrive = 1 (unity)y = tanh(k · x) · 0.9 + 0.05 · x + 0.03 · x² · sgn(x)k = 1 + 1.8 · drive1 + 1.8 · drivedrive 0 → 1× (unity) drive 0.3 → 1.54× drive 1.0 → 2.8×out = dry + comp(band) − compIdle(band) → dryDynamic EQ sits between corrective static EQ and the main compressor, or acts as the last surgical stage on a vocal bus.
Compression grew out of the needs of radio and vinyl in the 1930s: signals had to be kept within a narrow level window or the transmitter would over-modulate and the cutter would jump the groove.
The first commercially successful limiter was the Western Electric 110A (1937); Bill Putnam Sr. built the first recording-studio compressor in 1959.
A compressor reduces the dynamic range of a signal — it turns down the loudest parts so the quiet parts can come up, making a performance sound even, controlled and loud.
A compressor maps input level to output level with a bend at the threshold. The five character buttons select true per-circuit DSP models — feed-forward (Digital, VCA, FET) vs feedback (VFET), peak vs RMS detectors, program-dependent ratio (FET) and program-dependent release (Opto) — each with a subtle, character-appropriate harmonic colour.
out = in, in ≤ thr | thr + (in − thr)/ratio, in > thrGR = in − out = (in − thr) · (1 − 1/ratio), in > thrdBFS = 20 · log₁₀( |x| )RMS = √( 1/N · Σ x² )out = comp(in) · 10^(makeup/20)On a channel, compress after corrective EQ so the compressor hears the cleaned signal.
The limiter is the extreme end of compression — a compressor with an infinite ratio and (ideally) zero attack, invented to protect transmitters and disc cutters from overload.
The “brickwall” concept — a ceiling no signal can cross — became essential in the 1990s digital loudness race.
A brickwall limiter prevents any sample from exceeding a set ceiling, so the output never clips.
It is used to maximize loudness by shaving only the loudest peaks, raising perceived loudness without distorting the body of the track.
A brickwall limiter enforces a hard ceiling: any sample (or inter-sample peak, in True-Peak mode) above the ceiling is pulled down to it. A limiter only ever reduces gain — never boosts.
out = min(in, ceiling) (per sample)GR = max(0, in − ceiling)TP = max over up-sampled peaks; ceiling − 0.3 dB if onLUFS = −0.691 + 10·log₁₀( mean of K-weighted 400 ms blocks² )The limiter is the LAST processor on the master bus, after EQ, compression and any colour.
Splitting the audio into frequency bands and compressing each separately was first done for broadcasting in the 1960s to keep vocals present over music; the Ward-Beck and Studer broadcast processors did it with analog crossovers.
In mastering, Bob Katz’s work on the K-system and units like the TC Electronic Finalizer (1996) made multi-band compression the loudness tool of the 1990s CD era.
A multi-band compressor applies different compression to low, mid and high frequencies separately, so a loud bass note does not pull down the vocal, and a bright cymbal does not duck the body.
A multi-band compressor splits the spectrum with Linkwitz-Riley crossovers, compresses each band separately, then sums the bands back. LR4 (24 dB/oct) crossovers sum flat at the crossover frequency.
LP = 2nd-order lowpass², HP = 2nd-order highpass² → −6 dB at f₀, sum = flatband_i = comp_i( LR_split_i(input) )out = Σᵢ band_i · 10^(makeup_i/20)Use it on the master bus AFTER EQ and single-band compression but BEFORE the final brickwall limiter, or on a drum/bass group bus.
Saturation is the “good” distortion every analogue device added: vacuum tubes, transformers and tape all bent the signal slightly as it passed through, adding harmonics and soft compression.
The warm sound of 1950s–60s recordings (RCA, Abbey Road) IS tube and tape saturation. When digital arrived clean in the 1980s engineers missed it — so plug-ins recreated the harmonic characters of tube, tape, transistor and opto circuits as a deliberate colour tool.
Saturation adds harmonics and gentle level-dependent compression, making a sound warmer, fatter, brighter and louder — and more “present” in a mix.
Every mode is a different waveshaper y = f(drive, x). The drive is applied twice — first as a linear pre-boost into the shaper, then again inside the curve — so the effective tanh drive squares. That double drive is what gives this panel its strong, harmonic-rich character. (The Dynamic EQ per-band saturator uses the same curve with 0.2× the coefficient and no pre-boost — see its notes.)
g = 1 + 9 · drivek = 1 + 9 · drivey = tanh(k · x) · 0.9 + 0.05 · x + 0.03 · x² · sgn(x)tanh(k·x·0.85) · (1−0.08|x|) / sgn(a)·(1−e^−|a|·2.5)·0.95 / 0.45x+0.55·tanh(k·x·0.5) / 0.7x+0.3·tanh(k·x)≈ (1 + 9 · drive)²drive 0 → 1× (clean) drive 0.4 → ≈21× drive 1.0 → 100×On a channel, saturate after compression to add excitement, or before it to feed the compressor a hotter signal.
Clipping is the hardest form of saturation — when a signal exceeds a circuit’s headroom the top of the waveform is simply chopped off.
Engineers first did this deliberately on tape and consoles to get loud, aggressive sounds (the “driven” rock vocal, the clipped snare), and clipping became a signature of heavy genres.
A clipper cuts the loudest peaks off the waveform, raising loudness more aggressively than a limiter because it does not try to hide the gain reduction — it simply removes the peak.
The clipper is an asymmetric soft-clip. Drive pushes the signal harder into a tanh curve; symmetry offsets the positive/negative halves to add even harmonics. Subtracting the DC term keeps the curve centred.
y = tanh( drive · (x + s) ) − tanh( drive · s )s = symmetry / 100 ( −1 … +1 )drive = 10^(driveDb/20)drive = 1, s = 0 → y = tanh(x) (clean soft-clip)On a drum bus or master, place a clipper BEFORE the final limiter to shave the sharpest peaks so the limiter does not have to clamp them (cleaner loudness).
Magnetic tape recording arrived with the German Magnetophon (1940s) and came to studios after WWII via Bing Crosby’s Ampex Model 200 (1947).
For forty years everything was recorded to analog tape, and its sound became “the sound of a record.” As digital took over in the 1990s the tape character became a desirable effect, recreated by plug-ins so a clean digital recording can be given an analog soul.
A tape simulator adds the analogue character of recording to magnetic tape: gentle saturation that compresses and warms the signal, high-frequency loss that tames harshness, a low-frequency “head bump” that adds weight, plus the pitch wobble (wow & flutter) and hiss of a real machine.
Tape colour comes from three stages in series: a saturation curve (tanh) for harmonic compression, a head-bump low shelf for bass weight, and a speed-dependent high-frequency loss that tames the top.
y = tanh( (1 + drive·8) · (1 − bias·0.5) · x )lowshelf, fc ≈ 90 Hz, gain = headBump · 10 dBlowpass fc = 18k / 12k / 8k (30 / 15 / 7.5 ips) − hfLossdelay = D + wow·sin(2π·0.7t) + flutter·sin(2π·6t)Use tape at the start of the chain or on a bus to commit a sound to “tape” early (like tracking to analog), or on the master after EQ/dynamics for final glue.
Delay began as a physical phenomenon — tape echo. Sam Howlett and later Charlie Watkins (Watkins Copicat, 1958) created the first tape-echo units by routing audio through a loop of magnetic tape with a movable playback head.
The Roland RE-201 Space Echo (1974) became the studio classic. Digital delay brought clean, long, modulated echoes, and the plugin era made tempo-synced, modulated delay a staple for vocals, guitars and electronic music.
A delay repeats the signal after a set time, with each repeat quieter (feedback) so the echoes decay.
A feedback delay feeds the output back into the input, so each echo is the last echo multiplied by the feedback. The echo train decays geometrically.
y[n] = x[n] + feedback · y[n − D]a_k = feedback^kD = (60 / BPM) · note (1/4 = 1, 1/8 = 0.5, dotted = ×1.5)Insert delay on a channel or, better, on a parallel send so you can keep the dry signal untouched.
Reverberation is the natural sound of a space — the many reflections that reach the listener after the direct sound.
Studios first captured it in echo chambers (EMI Abbey Road, Capitol Studios, 1950s) and mechanical plates and springs. Digital reverb arrived in the 1970s, inventing algorithmic reverb, and convolution reverb let us sample real halls.
Reverb adds a sense of space, distance and depth, placing a dry sound in a virtual room, hall or plate.
This reverb is convolution-based: a synthetic impulse response (decaying noise) is convolved with the signal. The tail envelope and damping filter set the room character.
env(t) = (1 − t/T)² · noise(t)y[n] = Σ_k h[k] · x[n − k]lowpass fc = 500 + damping · 19500 Hzlength = sampleRate · decay · (0.7 + size·0.6)Put reverb late in the chain — after EQ, dynamics and saturation — so you reverberate the finished sound, not the problems.
Mastering began in the vinyl era: the “transfer engineer” cut a lacquer from the final mix, balancing levels and protecting the cutter from overload.
As formats multiplied (cassette, CD, streaming) mastering became a dedicated art — the last creative and quality-control step, handled by specialists like Bob Ludwig and Emily Lazar.
Mastering is the final polish of a finished mix.
Mastering balances the spectrum and reaches a target loudness. The K-system defines reference headroom (0 K-meter = a fixed dB below 0 dBFS), and streaming platforms normalise to roughly −14 LUFS.
K-20: 0 K = −20 dBFS K-14: 0 K = −14 dBFS K-12: 0 K = −12 dBFS≈ −14 LUFS integrated (Spotify / Apple normalize to this)ceiling ≤ −1 dBTP for lossy encodes (codec overs)Mastering is the FINAL stage — run it on a finished, approved mix, not a work-in-progress.
Section mastering is the art of treating different parts of a track — intro, verse, chorus, outro — with different processing, because a single static master is a compromise across material that changes in level, tone and energy.
On the analog console this was done with automation — riding levels, EQ and limiter thresholds across the song — or by mastering each section separately and editing the results together. In the box, parameter automation solved the level side; the DSP that morphs between settings in real time is the modern extension.
Section mastering lets one track be mastered for its own dynamics — a quiet intro can stay open and dynamic while the chorus is pushed loud, with a smooth morph between them instead of an audible snap.
Across a glide zone of half-width h around cue c, the playhead position t (0 at the left edge, 1 at the right) blends every parameter between the two adjacent preset recipes. EQ and Compressor interpolate parameters using a neutral (flat / ratio-1) state when a side has the module off, so the effect ramps in/out. Limiter, Tape and Saturation scale their wet/dry mix by each side's presence. Multi-band uses a continuous crossfade weight (0..1) so it passes through with no graph rewire.
t = (frac − (c − h)) / (2h), 0 ≤ t ≤ 1h = glide · min(leftSpan, rightSpan, 0.18)left = 1 − t, right = t (at edges t=0/t=1 the blend = pure preset)g = g_A·(1−t) + g_B·tmix = mix_A·(1−t) + mix_B·t (ON side's params)w = (onA ? 1−t : 0) + (onB ? t : 0) → engine crossfade, no rewireat zone edges the blend = pure preset → continuous, no audible snapUse Section Mastering when one static master cannot serve the whole track — a dynamic intro that needs to breathe and a loud chorus that needs to hit.
The waveform of a finished master has always arrived too late. On vinyl you saw the groove after the lacquer was cut. On tape you saw the trace after the pass. In the DAW you see the source file the moment it is loaded — but the mastered file only after you bounce, render, or export. Until then the work has no picture of itself.
Meters tell you what is happening in this buffer. The source overview is a picture of a file that already exists on disk. Neither is an image of the master as a whole, because recording culture decided that proof of a finished sound-work is storage.
Mastered Output inverts that order. It draws the full duration of audio that has been through the live chain but has not been written to any path. No bounce. No object in the filesystem. You are looking at how the master would look — an image of a file that does not exist yet.
That idea is Dr Ivan Zavada’s. Senior Lecturer and Program Leader in Composition & Music Technology at the Sydney Conservatorium of Music, The University of Sydney, he is a composer and digital media designer whose work examines the relationship between concrete sounds on a fixed recorded medium and visual elements of abstraction. Mastered Output is that question made into a studio view: a picture of a sound-work before it has been inscribed. Record: https://profiles.sydney.edu.au/ivan.zavada
This is not a courtesy preview and not a render waiting to finish. It is a third kind of image: neither the living meter nor the stored source, but the would-be file — the whole duration, as it will be if you export right now, while the thing itself still only exists as process.
The motto of Sound Chain Master is The best master is yet to come… In the studio it means the bounce is still ahead — you are looking at a master that has not been born. The same words carry an older hope: that the finest has not yet arrived, that the Master is still coming. Those layers are not a pun. Sound has always lived in all of them at once — as craft, as vibration, as listening, as devotion, as a signal on a chain. A picture of a file that does not exist yet is the studio form of that hope: the work is still becoming.
Once you take that seriously, the faders are not cosmetics on a graph. Pull Limiter Out and you are editing an artifact that has not been born. Move the automation lane and the future object changes its body in time. Seeing the master before it becomes a file is the innovation — created and designed by Ivan Zavada (Spher8 · SCM), © 2026.
The VU (Volume Unit) meter was standardised in 1939 by a joint NBC/CBS/Bell Labs committee to give broadcasters a consistent reading of programme loudness.
Its d’Arsonval needle movement has a defined 300 ms rise (the “VU ballistic”), and 0 VU was calibrated to a reference level. The Weston and Simpson VU meters became the studio standard for forty years.
A VU meter shows the perceived loudness of a signal, not its peaks — its slow needle averages the level the way the ear hears it.
It is used for gain staging.
Use a VU meter at the START of the chain to set input gain, and at the end to confirm output level.
Peak meters replaced VU meters as digital recording needed sample-accurate overload detection: a single sample over 0 dBFS clips, something the slow VU needle cannot show.
LED-segment meters and digital ppm/peak meters became standard in the 1980s–90s. Modern meters show peak, RMS, headroom and true-peak (intersample) so an engineer can both avoid clipping and judge loudness.
This segmented meter shows the peak level of the signal in dBFS so you know exactly how close you are to clipping (0 dBFS).
In PRO mode it adds RMS, per-channel peak and headroom, letting you gain-stage accurately and confirm the master leaves safe headroom.
LUFS (Loudness Units relative to Full Scale) is computed in four steps: K-weighting → mean square → time-window average → log. The integrated measure adds a gating stage that ignores silence.
H₁(s): high shelf, fc = 1681.8 Hz, gain ≈ +4 dBH₂(s): 2nd-order Butterworth HPF, fc = 38.1 Hzz_i = (1/T) · ∫ |K(x_i(t))|² dt over window TL = −0.691 + 10 · log₁₀( z_L + z_R ) [LUFS]T = 400 ms, rectangular sliding window, hop = 100 msT = 3000 ms, rectangular sliding window, hop = 100 msT = ∞ (gated: ignore blocks where L < -70 LUFS, then < Γ − 10 LU)Upsample ×4, find max |x|, convert to dB — catches inter-sample oversLUFS Time Windows — Momentary / Short-Term / Integrated
Watch the level meter at the END of the chain (OUT mode) to confirm the master never clips and leaves headroom, and at the input (IN mode) to set source gain.
Spectrum analyzers entered studios with the Kay Elementrics and Hewlett-Packard analyzers of the 1960s–70s and the dedicated RTA (real-time analyzer) used in live sound and mastering.
The oscilloscope is older still — cathode-ray waveform viewing from the 1930s. In the plugin era the combined spectrum + scope + loudness meter became the default “what does my master actually look like” display for mastering engineers.
This visualizer shows the spectrum (level per frequency), the waveform (oscilloscope), and a live peak/RMS readout of the final output, plus a ghost trace of the dry pre-chain source so you can A/B the mastered signal against the original on the same analyzer.
Read this at the END of the chain (it taps the post-master analyzer).
The concept of routing a signal through a fixed order of processors — the “signal chain” — comes from the analog studio, where a microphone, preamp, EQ, compressor and tape machine were wired in sequence on a patchbay.
The order mattered then and it matters now: each processor feeds the next, and moving one changes how they interact.
The Audio Source loads a file or your microphone, plays it through the chain, and gives transport, output volume, a global bypass and a reset.
It is the entry point of the whole signal chain — everything below processes whatever you feed it here.
This is the START of the chain — load or mic your source here, set a sensible output level, then build your effect chain below.
The signal chain is the spine of every record. In the analog studio a signal travelled a fixed physical path — microphone → preamp → EQ → compressor → tape machine — wired in sequence on a patchbay, and the order was set by which sockets you patched.
Moving a processor in the chain changed how it interacted with the next: a compressor before an EQ reacts to an un-shaped signal; an EQ before a compressor lets you de-ess before it gets grabbed. Parallel routing is just as old.
The order of processors is as important as the processors themselves. The same EQ and compressor sound different in different orders, and effects like reverb and delay are often better in parallel so the dry signal stays untouched.
Think of the chain in three stages: corrective → dynamic control → colour & space → level.
LUFS (Loudness Units relative to Full Scale) was standardised in ITU-R BS.1770 (2006) and adopted by the EBU as EBU R 128 (2010). It replaced the competing — and often misleading — peak and RMS loudness metrics that drove the loudness war of the 1990s–2000s.
LUFS measures perceived loudness — not peak amplitude — using a K-weighting filter that models how the human ear hears. Three time windows serve different engineering purposes.
LUFS is computed in four steps: K-weighting → mean square → time-window average → log. The integrated measure adds a gating stage that ignores silence.
H₁(s): high shelf, fc = 1681.8 Hz, gain ≈ +4 dBH₂(s): 2nd-order Butterworth HPF, fc = 38.1 Hzz_i = (1/T) · ∫ |K(x_i(t))|² dt over window TL = −0.691 + 10 · log₁₀( z_L + z_R ) [LUFS]T = 400 ms, rectangular sliding window, hop = 100 msT = 3000 ms, rectangular sliding window, hop = 100 msT = ∞ (gated: ignore blocks where L < -70 LUFS, then < Γ − 10 LU)Upsample ×4, find max |x|, convert to dB — catches inter-sample oversLUFS Time Windows — Momentary / Short-Term / Integrated
Watch Short-Term while mixing; deliver to an Integrated LUFS target for your platform.
Stereo imaging is the art of controlling where sound sits between the left and right speakers — its width, its centre, and its place in the soundstage.
It grew out of the Mid/Side (M/S) microphone technique invented by EMI engineer Alan Blumlein in the 1930s: a cardioid Mid mic captures L+R, a figure-8 Side mic captures L−R, and the two are matrixed back to stereo. That same matrix is the engine of every modern imager.
An imager reshapes the stereo field — making a mix wider or narrower, collapsing the bass to the centre, rebalancing a lopsided image, or gently rotating the stereo perspective.
The imager works in the Mid/Side domain. The stereo signal is encoded to Mid (the centre) and Side (the width), the Side is scaled by the width factor(s), then everything is decoded back to L/R. Direction applies a rotation matrix on the final L/R.
M = (L + R) / 2, S = (L − R) / 2L = M + w·S, R = M − w·SS_out = wLo · LP(S, fc) + wHi · HP(S, fc)L' = cosθ·L − sinθ·R, R' = sinθ·L + cosθ·Rw = 1, θ = 0 → L = M + S = L, R = M − S = RL = R = M → fully monoUnderstand width, balance and mono compatibility at a glance
A tight vertical shape means the signal is mostly centered and mono-compatible.
A wider, symmetrical shape usually means healthy stereo width with good left/right balance.
If it spreads too far sideways, the mix may lose focus or collapse in mono.
Messy sideways or inverted-looking readings can point to phase problems and weak mono playback.
If the reading leans left, the stereo image is unbalanced toward the left channel.
If the reading leans right, the stereo image is unbalanced toward the right channel.
Quick takeaway: A good stereo image is not just wide. It should also be balanced, focused and mono-safe. Use your ears first, then let the vectorscope confirm what you hear.
Put the imager late in the mastering chain — after EQ and dynamics, before or just before the limiter — so it shapes the finished sound.
A studio's look is part of its identity — the teal of a Neve console, the amber LEDs of a tape machine, the green faceplate of a Brüel & Kjær analyser. Plugin designers have long offered skins so the same DSP can wear several studio personalities.
Sound Chain Master ships six full visual identities, each a complete token set — page background, ambient glow, logo and title gradients, accent colour and a signature overlay (frosted glass caustic, brushed-metal grain, CRT scanlines, a HUD grid) — so the panels read as materially different instruments, not just recoloured chrome.
The identity selector re-skins the whole studio in one click; the panel-layout selector controls how many panels sit side-by-side; and the display mode switches the viewing chrome.
Pick an identity to match the session's mood or the material, then choose a panel layout for the screen width.
Low frequencies carry almost no directional cue — the human ear localises bass poorly, and stereo bass is summed to mono on nearly every playback system (phones, Bluetooth, club PAs, radio).
Bass-mono processors — Tone Projects Basslane (2008), bx_shredspread and Ableton's Bass Mono — were built to collapse the low end to the centre before that summing happens, so the bass stays full and phase-coherent everywhere instead of thinning or cancelling in mono.
Mon8 mono-izes (or partially narrows) only the low-frequency content below a chosen cutoff, leaving the mid and high frequencies in their original stereo field.
Processing is done in the Mid/Side domain. The Side (difference) signal is high-passed at the cutoff; the Mid (sum) is untouched. The Width dial blends the full-band Side with the high-passed Side.
M = (L + R)/2, S = (L − R)/2S_HP = HPF(S, fc)S' = w·S_HP + (1 − w)·S (w = 1 → mono below fc)L' = M + S', R' = M − S'Place Mon8 early in the chain — before stereo widening and the limiter — so the bass is centred before it is widened or limited.
The Black Box Analog Design HG-2 is a hardware tube density processor built around custom input/output transformers and a 6U8A valve run in both pentode and triode configurations, with a parallel 12AX7 saturation path.
Brainworx modelled the unit in software, adding digital-only controls (Density, Input Gain, Calibration, Air Amount, Mix) that make the hardware’s “make it louder without raising peaks” trick precise and repeatable.
Analogue Density adds rich, musical harmonics, natural soft-knee compression and transformer warmth while staying transparent at low drive — it makes material sound denser, fuller and more “expensive” at the same peak level.
Density is bipolar: d ∈ [−1, +1] (dial %÷100). Positive d pushes drive into both tubes and attenuates the output to hold loudness; negative d subtracts drive (relaxes the tubes). The Pentode and Triode knobs add base drive that Density then pushes or pulls relative to.
push = max(0, d)·0.55, pull = max(0, −d)·0.25drive = max(0, (pentode + push − pull)·0.6)out = 1 / (1 + max(0, d)·0.85·0.6) (positive d only)no push, no comp — only subtracts drive → cleans/softensM = (L + R)/2, S = (L − R)/2drive_M = mapping(midDensity), drive_S = mapping(sideDensity)curve set for max(drive_M, drive_S); softer channel scaled by its ratioL' = M + S, R' = M − SPlace Analogue Density in the pre-master, after corrective EQ and before the limiter — it is a colour / density stage, not a level stage.
Spher8 · SCM — Sound Chain Master · Educational Booklet