The Audio Processors
Every audio channel strip in AnalogDAW carries analog-modeled processors. This chapter explains each one in plain language — what it does, when to use it, and what every control means. The same processors appear (in slightly different combinations) on tracks, mix buses, aux returns, and the master; see Channel Strip Types for which strip has which.
Everything is musical by default. Each processor models the behavior of classic analog hardware, so even pushed hard it tends to sound pleasing rather than harsh. Most processors are off until you enable them, except the preamp, which is always shaping the sound.
Ordering & enabling processors
The processors on a strip form a chain — signal flows through them in order, top to bottom. That order is yours to change: select the stacked-layers Processing Order button beside the strip's DYN / EQ / INS / SND controls, then drag a row to reshape the signal path. Drop on the upper or lower half of a row to place the stage before or after it. Reset to Default restores the strip's original chain. For example, put the compressor before the EQ to even out a level before you sculpt tone, or drop an amp-sim insert ahead of the preamp. The order is saved with the session and restored exactly when you reopen it.
The row icons distinguish active, bypassed, empty, unavailable, and fixed stages. VoiceOver and other assistive-technology users can use each row's Move Up and Move Down actions. Processing Order is locked during count-in and recording so the capture path cannot change halfway through a take.
Track chains also contain a movable Record Point. Audio flows from top to bottom: every enabled processor above Record Point is committed to a new recording, while every processor below Record Point stays non-destructive and works during monitoring, playback, and export as before. Record Point cannot be deleted. New tracks—and sessions created before this feature—put it at the top, preserving dry recording by default.
Printing is irreversible for that take, so use it when you want to commit. Two common examples are putting an amp simulator before Record Point to record the finished guitar sound, or putting a network receiver there to record its remote audio. Move Record Point back above the processor for the next dry take.
Turning a built-in processor off doesn't just mute its effect — it is removed from the signal path entirely, so a disabled EQ, gate, compressor, de-esser, or filter costs zero processing power. (Plug-in inserts and the master limiter are the exception: when bypassed they stay in place and pass audio through untouched, so you can A/B them instantly.) Because a built-in toggle re-routes the audio graph, expect a brief moment of silence when you switch one on or off during playback — this is normal.
The preamp is always in the path (see below); you can switch off its coloring for a clean pass-through, but its pad and phase controls still apply.
Preamp & Color
The preamp is the front end of every channel — the analog gain stage your signal hits first. It's always in the path (you can switch its coloring off for a clean pass-through).
| Control | Range | What it does |
|---|---|---|
| Preamp Enable (PRE) | on/off | Switches the preamp stage in or out. See the note below — "off" is the only truly unaltered path. |
| Gain (GAIN) | −20 to +20 dB | Input gain into the channel and its harmonic stage. Drive it for more color, back it off for clean. |
| Trim (TRIM) | −20 to +20 dB | A clean level trim after the saturation, to set a consistent output without changing the color. |
| Color Enable (CLR) | on/off | Toggles the harmonic bloom. Turning it on restores your last Drive amount; turning it off sets Drive to 0. |
| Drive (DRIVE) | 0 to 1 | The amount of transformer-style harmonic "bloom." 0 is a clean output stage; higher adds rich harmonics. It's level-compensated, so it adds warmth without making the channel louder. |
| Tilt (TILT) | −1 to +1 | The tone of the color. Blue (left) emphasizes low/low-mids for weight; Red (right) emphasizes high-mids/highs for air and edge. |
| Phase (Ø) | on/off | Flips the polarity of the signal — useful for multi-mic sources (e.g. top/bottom snare) that fight each other. |
| Pad (PAD) | on/off | A −20 dB input pad ahead of the gain stage, for very hot sources that would otherwise overload. |
Use it for: setting healthy levels, and adding analog warmth or bite. A little Drive on a sterile digital recording can make it feel "recorded to a console."
Even at COLOR 0 and GAIN 0, an enabled preamp is not a clean wire. The COLOR knob adds an extra bloom on top (which is why 0 is described as a clean output stage), but the preamp still imparts a baseline console character — gentle voicing EQ plus transformer and Class-A harmonic warmth — whenever it's enabled. To get a truly unaltered track, switch the preamp off (its pad and phase still apply). See The Console Sound for the full picture of what's baked in where.
Filters (High-Pass / Low-Pass)
Simple but essential frequency filters to remove unwanted extremes.
| Control | Range (default) | What it does |
|---|---|---|
| High-Pass (HPF) | 20 to 600 Hz (off, 80 Hz) | Removes low frequencies below the set point — rumble, plosives, mud. Raise it to clean up the low end of vocals and guitars. |
| Low-Pass (LPF) | 2 to 20 kHz (off, 18 kHz) | Removes high frequencies above the set point — hiss, harshness, air. |
Each filter has its own on/off switch. The ranges are deliberately asymmetric — these are corrective filters, so the low-cut stays in the low end and the high-cut stays up top rather than sweeping across the whole spectrum.
Use it for: cleaning tracks before they hit the EQ and compressor. A high-pass on everything that isn't bass or kick is a classic mix move.
Gate
A noise gate automatically quiets a track when the signal falls below a threshold — it "closes" during silence and "opens" when you play.
| Control | Range | What it does |
|---|---|---|
| Threshold (THR) | −80 to 0 dB (−40) | The level the signal must exceed for the gate to open. |
| Range (RNG) | −80 to 0 dB (−60) | How much the gate turns down when closed. 0 = bypass, −80 = a hard, fully-silent gate. The default fades naturally rather than slamming shut. |
| Attack (ATK) | 0.1 to 200 ms (1) | How quickly the gate opens once the signal crosses the threshold. Fast for percussive sources. |
| Hold (HOLD) | 1 to 500 ms (50) | The minimum time the gate stays open after the signal drops below threshold — prevents chatter on a source that hovers around the threshold. |
| Release (REL) | 10 to 1000 ms (150) | How quickly the gate closes after the hold time. Longer = smoother, more natural tails. |
| Sidechain (SC) | track/bus/aux | Optionally open the gate from another track, group bus, or aux return instead of its own signal (e.g. gate a pad with a kick drum for rhythmic effects). |
Use it for: tightening drums, removing bleed and hiss between phrases, killing amp hum during silences.
EQ (Equalizer)
A musical 4-band parametric equalizer modeled on a classic transformer console. Use it to shape tone — boost what's missing, cut what's in the way. (High- and low-pass filtering is handled by the separate Filters stage above.)
The four bands and their default homes:
| Band | Frequency range (default) | Typical use |
|---|---|---|
| Low Shelf (LF) | 30 to 600 Hz (60 Hz) | Weight and "thump" in the low end. |
| Low-Mid (LMF), bell | 200 Hz to 4 kHz (360 Hz) | Body and warmth, or cutting boxiness/mud. |
| High-Mid (HMF), bell | 800 Hz to 12 kHz (3.2 kHz) | Presence, attack, and intelligibility. |
| High Shelf (HF) | 2 to 22 kHz (12 kHz) | Air and sheen up top. |
Each band is confined to its own region of the spectrum, the way a console's fixed-band EQ is. Each has:
- Frequency (Hz) — where it works, within the band's range above. The very top of the HF band is limited by your session's sample rate: at 44.1 kHz the shelf stops near 20 kHz however far you turn it, so the last of that knob's travel does nothing. Work at 48 kHz or higher to reach the full 22 kHz.
- Gain (dB) — −18 to +18 dB of boost or cut.
- Q — 0.3 to 6: bandwidth, where low Q is broad and gentle and high Q is narrow and surgical. For the shelves, Q controls the gentle console "bump" at the corner (0.7 is a flat shelf; the default 1.1 gives the characteristic console lift).
- An enable switch, and a Bell (BELL) switch on the two shelves that turns a shelf into a fully parametric bell for precise surgery.
Two console-style voicing options shape the whole EQ:
- Proportional Q (PROP Q) — on by default. A bell automatically widens at low boost and tightens as you push it, giving forgiving, musical curves. Turn it off for literal, surgical control. It affects the two bell bands only.
- Harmonics (BLOOM) — 0 to 1, default 0.35. Even-dominant Class-A saturation added whenever the EQ is enabled, for that "expensive console" sheen. 0 is a mathematically clean EQ.
Use it for: everything — corrective cuts to fix problems, and creative boosts to flatter a source.
Compressor
The compressor controls dynamics — it makes loud parts quieter so the overall level is steadier and more powerful. AnalogDAW's compressor is special: you choose a mode that models a different classic circuit, and the mode changes both the sound and the controls to match the real hardware.
The same five modes are available on tracks, buses, aux returns, and the master — so a "FET" picked on a bus genuinely sounds like a FET.
The five compressor modes
| Mode | Character | Reach for it on… |
|---|---|---|
| FET | Fast and aggressive. Adds punch and attitude. | Drums, vocals, bass, room mics. |
| OPTO | Smooth and transparent. Flattering, almost invisible level control. | Vocals, bass, anything you want gently leveled. |
| TUBE (variable-µ) | Warm and gluey, rich harmonics. | Mix bus, vocals, mastering. |
| VCA | Clean and precise. Surgical, punchy. | Snare, tight drums, anything needing exact control. |
| BUS | "Glue." Gels a whole mix or drum bus into one cohesive sound. | Drum bus, mix bus, master. |
Common controls
Underneath, every mode drives the same set of parameters — but each mode only shows the controls its hardware actually had, and interprets them its own way:
- Threshold (THR) — the level above which compression begins. The FET calls its equivalent Input (IN), and the OPTO folds it into Peak Reduction.
- Ratio (RAT) — how hard it compresses (e.g. 4:1 means 4 dB in → 1 dB out). Continuous on the VCA, a discrete selector on the FET and BUS, and absent on the OPTO and TUBE, which have fixed curves.
- Attack (ATK) — how fast it grabs the signal. Fast tames transients; slow lets punch through.
- Release (REL) — how fast it lets go. Set it to "breathe" with the music.
- Makeup / Output (MKUP / OUT / GAIN) — brings the level back up after compression.
- Knee — how gradually compression engages around the threshold. Only the VCA exposes it, as its KNEE switch; the other modes use the knee baked into the circuit they model.
Mode-specific controls
Each mode also exposes the real front-panel controls of its inspiration:
- FET — a Ratio (RAT) selector with the classic 4:1, 8:1, 12:1, and 20:1 positions, plus an All-Buttons-In ("British mode") position that pins the ratio to 20:1, softens the input drive, and slows the release for the famous aggressive, distorted smash. Its Input (IN) and Output (OUT) knobs are the level controls; Attack (ATK) and Release (REL) run on the FET's own 1–7 scale, where 7 is the fastest.
- OPTO — a single Peak Reduction (PK RED) knob plus a Gain (GAIN) makeup knob, and a Mode (MODE) switch with COMP and LIM positions. Attack and release are program-dependent (they react to the music) by design, so the opto exposes no attack or release control.
- TUBE — Input (IN), Threshold (THR), and Output (OUT) knobs, plus a Time Constant (TC) selector (positions 1–6, each a different attack/release pairing; 5 and 6 are program-dependent).
- VCA — continuous Threshold (THR), Ratio (RAT), and Output (OUT) knobs, a true-RMS detector, and a Knee (KNEE) switch: HARD grabs sharply at the threshold for punch, EASY eases in gradually for transparent control.
- BUS — discrete Ratio (RAT), Attack (ATK), and Release (REL) switches, including an Auto release position (program-dependent), plus Threshold (THR) and Makeup (MKUP) knobs. The AUTO pill under the Makeup knob engages auto make-up, which tracks the average gain reduction so the output stays near unity however hard the compressor works; while it is on, the Makeup knob is dimmed and ignores input. Auto make-up is available wherever BUS mode is selected — tracks, buses, aux returns, and the master.
Sidechain
The compressor can be triggered from another track, group bus, or aux return instead of its own signal — the classic example is ducking a music bed under a voice, or pumping a bass against the kick. Track, bus, and aux strips can use different sources for their gate and compressor; pick each processor's sidechain source independently.
Worked example — ducking a bass against the kick
- Select the Bass track and open its Compressor.
- Set Sidechain (SC) to the Kick track. The compressor now listens to the kick while still compressing the bass.
- Set Ratio around 4:1 and lower Threshold until the gain-reduction meter moves only when the kick hits — a few dB is usually plenty.
- Set Attack fast (1–5 ms) so the duck starts on the transient, and Release to roughly the gap between kicks (80–150 ms at most tempos) so the bass is back up before the next one.
Two things worth knowing about how sidechains behave here:
- The key is the source's pre-fader signal. Pulling the kick's fader down — or muting it — does not change how hard it ducks the bass. That is deliberate: it lets you use a track purely as a trigger. To silence a trigger track in the mix while still using it as a key, turn its output down rather than removing it.
- Sidechain loops are ignored. If A is keyed from B and B from A, AnalogDAW drops the edge that would close the loop rather than producing an undefined result. Chains (A ← B ← C) work normally, in order.
Use it for: evening out performances, adding punch, controlling peaks, and gluing groups of instruments together.
De-Esser
A de-esser is a specialized compressor that reacts only to sibilance — the harsh "sss" and "shh" sounds in vocals — and tames them without dulling the whole track.
| Control | Range | What it does |
|---|---|---|
| Threshold (THR) | −40 to 0 dB (−20) | How loud the sibilance must be before it's reduced. |
| Frequency (FREQ) | 3 to 12 kHz (7) | Where the sibilance lives (around 7 kHz is typical). |
| Range (RNG) | 0 to 20 dB (6) | The maximum amount of reduction applied. |
Use it for: vocals and voice-overs with piercing "S" sounds; also cymbals or acoustic guitars that are too bright in the highs.
Plug-in Inserts
Beyond the built-in processors, every track, mix bus, aux return, and the master has six insert slots for hosting Audio Unit (AU) plug-ins — the third-party effects you've installed.
- Load a plug-in into a slot from the Inserts (INS) section. The slot menu
groups your Audio Units into submenus by manufacturer so a long list stays
navigable; when you know the name, choose Search Plug-ins… to open a
searchable browser and type any part of the plug-in or manufacturer name to
filter (matching is case- and accent-insensitive, and multiple words all have to
match — e.g.
acme comp). - Open its editor to access the plug-in's own interface. The editor has two tabs: Plugin UI (the plug-in vendor's own interface) and Controls (a generic list of every parameter, handy when a plug-in has no custom UI). You can also open a plug-in's editor from a control surface.
- Close and reopen an editor freely — closing the window only hides it; the plug-in keeps running and its UI comes right back when you reopen it.
- Keep working from the editor on macOS. A plug-in editor is attached to its project window, opens within that window's bounds when its size permits, and follows it through moves, minimization, and full-screen Spaces. If a plug-in is larger than the project window, its title bar remains reachable. Keyboard shortcuts the plug-in does not handle — including AnalogDAW's transport shortcuts — continue to reach the project. Typing in a plug-in text field stays with the plug-in.
- Watch its meters. The editor you open is the same plug-in instance that processes your audio, so a plug-in's own gain-reduction meters, analysers, and level displays move during playback and while input monitoring — not just the channel strip's meters.
- Bypass a slot to compare with/without. The plug-in stays loaded and keeps its settings, so switching back and forth is an A/B of the same instance rather than a reload.
- Inserts live in the reorderable chain, so you can place them before or after the built-in EQ, compressor, etc. — and even ahead of the preamp, so an amp-sim or channel-strip plug-in can feed into the built-in preamp color.
A plug-in's own sound is heard while its editor is open. Some plug-ins make sound the project knows nothing about — previewing a note you click in a pitch editor, auditioning a sample from the plug-in's own keyboard. AnalogDAW keeps a plug-in's chain running with the transport stopped for as long as its editor is open, so those previews are audible. Close the editor and the slot goes fully idle again: that idle cost is real, which is why it is tied to the window being open — an insert nobody is looking at never pays it.
Hosted plug-ins are told where the transport is. Every insert and instrument receives the project's tempo, time signature, play/record state, cycle range, and the exact timeline position of the audio it is being handed, on every processing block. Tempo-synced delays and modulation lock to the project, and pitch editors that work by transferring audio — playing a part through the plug-in so it can capture and analyse it — file that audio at the right place on the timeline instead of receiving nothing. If you use one of these, insert it, arm its transfer, and play the section: what lands in the plug-in matches where it sits in the project.
Plug-in settings are saved with the session, so your projects reopen exactly as you left them (where the plug-in is still installed). State is stored with the project the way the plug-in expects, so instances that carry their own presets or licensing (e.g. amp modelers, UAD effects) restore cleanly. Inserts are restored as part of session open — you do not need to open the plug-in editor before the plug-in processes audio, reports latency, or becomes available to control surfaces. When you do open a third-party editor, AnalogDAW brings the idle render host online before requesting the vendor Plugin UI, then refreshes that UI against the already-restored AU parameters so it matches the generic Controls tab immediately.
If a plug-in fails to load, the slot is held in a safe failed state so it does not add stale latency or process silence. Reopening the session treats that failure as runtime-only and retries the plug-in automatically; use Retry Current Audio Unit from the slot menu when you want to retry immediately without reopening the project. AnalogDAW also bounds implausible latency and tail-time reports from third-party Audio Units so faulty metadata cannot create an unbounded delay-compensation buffer or export pre-roll.
Where plug-ins come from
- On macOS, AnalogDAW hosts both AUv2 and AUv3 effects — the Audio Units you've installed under Library › Audio › Plug-Ins › Components.
- On iPhone and iPad, plug-ins are AUv3 effects that ship inside apps you install from the App Store. After installing one, open its app once so the system registers it — then it becomes available to AnalogDAW (and every other host).
If you just installed a plug-in and don't see it yet, open the slot menu and choose Refresh Audio Units; relaunching AnalogDAW also rescans. AnalogDAW lists both standard effects and MIDI-controllable music effect plug-ins, so tempo-synced delays, filters, and the like all show up. Active inserts continue to receive silent render buffers while Core Audio is enabled and the transport is stopped. This lets network receivers and generator-style effects deliver audio without requiring empty timeline playback.
Plug-in DSP at small buffers
On macOS, AnalogDAW crash-isolates third-party effect Audio Units in a separate host process. That keeps a failing plug-in from taking down the project, but the process boundary adds a fixed cost to every audio buffer. At 48 kHz, a 64-sample buffer gives the complete mix only about 1.3 ms; 128 samples gives 2.7 ms. A compound amp-studio plug-in that runs an amp, cabinet, room, and several pedal or studio effects can therefore overload at 64 even when the Mac's overall CPU meter looks comfortable.
If the DSP badge is unstable, first try 128 or 256 samples, bypass unused sections inside the plug-in, and prefer mono processing before the amp when the plug-in offers it. Raise the buffer further while mixing. This changes realtime headroom, not the sound or the contents of recorded takes; the trade-off is higher live-monitoring latency.
The other lever — and the decisive one when it is the number of heavy plug-ins rather than any single one — is to freeze the tracks you have finished. A frozen track's plug-ins are not rendered and are unloaded, so they cost neither DSP time nor memory until you unfreeze. Ten frozen guitar tracks carrying an amp-studio plug-in each give all of that headroom back to the track you are still recording.
SonoBus on iPhone and iPad
Load SonoBus in a normal insert slot. SonoBus publishes its iOS AUv3 as a music effect, not as an AU instrument, so it does not appear in AnalogDAW's Instrument picker. That picker contains only plug-ins that publish an instrument component.
To record received SonoBus audio:
- Leave the track at No Input—this avoids activating the built-in mic.
- Load SonoBus in an insert, then open Processing Order and drag that insert above Record Point.
- Arm the track, join the remote group in SonoBus, and press Record.
Arming a No Input track is allowed only when it has an active insert before Record Point. AnalogDAW feeds that insert silence, keeps it rendering while the transport is stopped, and records whatever the plug-in produces. A conventional effect that requires input will simply produce silence in this setup.
After joining the remote group in SonoBus, open its Plugin UI and use SonoBus's own Unmute control if it shows the yellow feedback-protection warning. Use headphones when receiving and sending through the same device; that warning and mute state belong to SonoBus rather than the host insert's bypass button. Keep Core Audio enabled so the network insert keeps rendering while the transport is stopped; this output-only workflow does not open the microphone.
The insert keeps streaming when you leave AnalogDAW for another app, and Core Audio comes back on its own after a phone call, an alarm, or a trip to the Home Screen — you should never have to switch it back on by hand. If Core Audio is off, nothing renders and SonoBus stays silent no matter what its own UI says, so that is the first thing to check when a network insert goes quiet.
Use it for: reverbs, delays, specialty effects, metering, and any favorite third-party processors.
Sends (Aux & Cue)
Sends tap a copy of the track's signal and route it somewhere else, while the track keeps playing through its own output.
- Aux sends feed aux return buses — used for shared effects like reverb and delay, so several tracks share one reverb. Each send has a level, an on/off, and a pre/post-fader switch (pre-fader is independent of the track fader; post-fader follows it).
- Cue sends feed cue buses — independent headphone mixes for performers, so the singer can hear more of themselves without changing the main mix. Like aux sends, each cue send has its own level and a pre/post-fader switch.
Sends are covered further in Mixing and Channel Strip Types.
Next: Recording Audio →