User Manual
Clear Voice LIVE
AI · Voice · Processor
Version 1.1.6
Made in Graz, Austria
© 2026 Fabio Schurischuster
1. Welcome
Clear Voice Live is a real-time AI voice processor designed for live sound,
broadcasting, conferences and streaming. It does four things at the same
time, on a single insert per vocal channel, at zero output latency in
ZERO mode (the default):
- Denoising — removes background noise, fans, HVAC, audience bleed
- Feedback prevention — kills mic-monitor coupling before resonance builds up
- Dereverb — tightens room reflections and hall tails
- Voice presence (called Debleed on the website) — keeps the spoken/sung voice itself untouched while pushing down bleed from drums, monitors and nearby instruments on the vocal mic
All processing is driven by an AI core that separates voice from background
frame by frame. There are no notch filters to draw, no rings to chase, no
static EQ.
The most important thing to know: where Clear Voice Live works best.
STRENGTH is the one fader that decides how the plug-in behaves. The colour scale
next to the fader (also in the GUI overview below) marks the working ranges:
- 0 to 2.9 - very light processing, mainly useful for a quick
before/after check.
- 3 to 5 - the normal working range. At 3 a
typical setup gets a gentle denoise and soft feedback prevention. If you need more, move towards
5.
- 5 to 8 - strong intervention. Use it when the situation calls
for it and listen carefully; the processing becomes audible.
- 8 to 10 - emergency range only: maximum suppression for a
situation that would otherwise be unusable.
DEREVERB works only while STRENGTH is above 0. At Strength 0 the whole
processing chain is bypassed, so Dereverb has no effect until you raise Strength.
The second setting that matters: pick the right CPU mode for the
job. In
feedback-heavy setups (loud monitors,
handheld mics near the PA) use
LOW — it is the most
stable, safest mode against acoustic coupling. For the
finest, best-sounding denoising (broadcast, streaming,
studio-grade cleanup) use
HIGH. Details in
section 7.
Voice only — everything else is treated as noise. Clear Voice
keeps the human voice (speech or singing) and reduces everything
else — including background music and instruments. It does
not tell "wanted music" apart from "unwanted noise": anything that is
not a human voice is treated as noise and suppressed. So if your goal is to
keep an instrument in the signal — e.g. a solo guitar played alongside
the voice — Clear Voice is not the right tool, because it will pull the
instrument down together with the noise. It is built for a clean, isolated
voice; an instrument-friendly version may come in the future.
New in 1.1 — trained for singing and speaking. Long
sustained sung notes, vibrato and falsetto are recognised as voice
and kept intact — while real feedback tones are still suppressed.
Clear Voice Live now works as well on a lead vocal as it does on speech.
Coming from v1.0? Expect more output level. The new
engine no longer trades level for cleanup — where v1.0 audibly pulled the
overall level down, v1.1 keeps the voice at full strength.
With identical rig settings you will get a hotter, more present
signal out of v1.1 — that is intentional, and it also means more
usable gain before feedback at the same perceived loudness. Re-check your
gain staging once when you switch a show file over.
2. Quick Start
- Install via the supplied
ClearVoiceLive-Setup-1.1.6.exe (Windows) or .pkg (macOS). v1.1 installs alongside an existing v1.0.0 — your old version stays exactly where it is.
- Re-scan plug-ins in your DAW. The new version appears as Clear Voice Live 1.1 under the Dynamics / FX category. If you already own v1.0.0 you will now see two entries — that is intended.
- Insert it on a single vocal channel — one instance per microphone
(see Best Practices).
- CPU: LOW. Latency: ZERO.
Strength: 0. Dereverb: 0.
- Increase Strength until the noise floor disappears without
losing voice character. For the finest denoising on an important channel,
switch CPU to HIGH — in feedback-critical monitor
setups, stay on LOW, it is the most stable choice
against coupling.
- Bring up Dereverb to taste to dry up the room.
That's the whole quick-start. The plug-in is intentionally
small — two faders, two safety buttons, two CPU modes, two latency modes.
Everything else in this manual is detail you only need when you want to
fine-tune.
💡 The four tips that matter most:
- First insert on the vocal channel, before the compressor.
Clear Voice Live wants the raw microphone signal; a compressor placed
after it smooths the gain moves (see
Best Practices).
- LOW CPU in ZERO latency is the right setting for about 90 % of
applications. It is the most stable mode against acoustic
coupling, it runs on every channel of the show, and it is what we tune
and test first. HIGH can sound a touch finer; if you want that, use it
in LOW latency, where it costs no more than LOW (see
section 7). In ZERO latency HIGH costs a
multiple of LOW, so reserve it for the one or two channels that earn it.
- Dereverb is feedback prevention, too. Room reflections
are part of every feedback loop. A small Dereverb (1 – 3) takes them
out of the signal before they can build, so the setup decouples more
reliably and holds more gain. Run it even on a dry stage (see
Dereverb).
- One instance per microphone. Insert Clear Voice Live
on each vocal channel rather than on a subgroup or bus — the model is
made for one voice at a time (see
Best Practices).
3. System Requirements
| Operating system | Windows 10 / 11 (64-bit). macOS 11+ (Universal: Apple Silicon & Intel). |
| CPU | x86-64 with SSE3, or Apple Silicon (native). |
| Plug-in formats | VST3 (Windows & macOS), AU (macOS). No AAX version —
Pro Tools users: Pro Tools loads AAX only; to use Clear Voice Live there,
load it through a VST3 wrapper such as Blue Cat PatchWork or DDMF Metaplugin. |
| Dependencies | None for the plug-in. Since v1.1 it is fully
self-contained: the AI model and all code ship inside the plug-in binary itself, the
C runtime is linked statically, and no extra DLLs or model files sit next to it.
(The Windows installer still carries the Microsoft VC++ runtime and installs it when
your system lacks it — a safety net for older Windows builds and for v1.0, not
something v1.1 needs.) |
| Host sample rates | 48, 96 and 192 kHz accepted. Other rates show a warning banner and bypass cleanly. |
| Internal processing | Always at 48 kHz, regardless of the host's sample rate (see section 4). |
| RAM | Under ~35 MB per instance — ten instances measured at 347 MB total host memory. The AI model ships inside the plug-in binary. |
| Output latency | 0.0 ms in ZERO mode (default). The optional LOW latency mode adds a fixed 127 samples = 2.65 ms, reported to the host as PDC (see section 7). |
The plug-in is written so that the host can route mono or stereo channels
freely. Stereo input is processed in stereo, with stereo-linked gain
decisions to keep the image stable.
4. Audio Setup & Buffer Size
In ZERO mode (the default) Clear Voice Live's
internal latency is zero and the plug-in itself adds nothing; in
LOW latency mode it adds a fixed 127 samples
(2.65 ms), reported to the host as PDC. The real latency you hear in a
live performance — the so-called roundtrip latency — is
determined by your audio interface, the host's buffer size, and the driver
running underneath.
Roundtrip ≈ (driver IN) + (host buffer) + (plug-in chain) + (driver OUT)
Internal sample rate
The AI core inside Clear Voice Live runs at 48 kHz fixed
for all hosts. When the host runs at 96 or 192 kHz the plug-in does
internal sample-rate conversion before and after the analysis path. The
output stays at the host's rate. This means:
- There is no audio-quality benefit to running the host
at 96 or 192 kHz just for this plug-in — the AI work
happens at 48 either way.
- There IS a latency benefit, because the audio driver's
buffer is measured in samples, not milliseconds. A 64-sample
buffer at 96 kHz is 0.67 ms; the same 64 samples at 48 kHz
is 1.33 ms. Running the host at the highest rate the interface
supports comfortably halves driver latency.
Recommended driver setup
| Setting | Recommended | Why |
| Audio driver | ASIO (Windows), Core Audio (Mac) | Both are low-latency, professional drivers; avoid WASAPI / WDM in live use. |
| Host sample rate | The highest rate your interface runs without complaints — usually 96 kHz, sometimes 192 kHz. | Reduces driver latency for the same buffer-size setting (samples ÷ rate). |
| Buffer size | 32 – 64 samples | The smallest your interface and CPU can handle without crackles. See below. |
Choosing the buffer size
Clear Voice Live is a live plug-in. Anything above ~64 samples
introduces enough roundtrip latency that the artist can hear themselves
twice through the in-ear monitor — once acoustically through the head,
once electrically through the chain. That feels wrong and is unworkable
for live use.
- 32 samples: tightest possible roundtrip. Use it if
your interface and CPU support it without dropouts. Best feel for the
artist.
- 64 samples: comfortable safety margin, still under
~3 ms roundtrip on a typical 96 kHz interface. This is the
sweet spot for most live shows.
Rule of thumb for live voice: pick the smallest buffer
size that runs your busiest show without dropouts on your hardware.
Higher rate + small buffer beats lower rate + medium buffer for live
feel.
Test before the show. If you haven't run Clear Voice Live
on the channel count you'll have on stage, reduce the show's risk by
doing a 10-minute soundcheck with all channels armed. CPU-related
dropouts usually appear at the moment everything talks at once.
5. GUI Overview
The plug-in window is intentionally small and resizable. Everything you
can change is on screen, no hidden tabs or sub-pages.
The plug-in interface — two faders, the key column (LINK, CPU, LATENCY, BYPASS, DELTA), three meters, the TRACK display and the PRESETS button.
Header
- Title plate — click anywhere on it to open the about overlay (legal info, license attributions, version).
- TRACK display — a segment display in the header centre showing which channel this instance sits on. It automatically abbreviates the track name your DAW provides (e.g. "BV1"); if the host doesn't pass a name, it shows the instance number. In the about overlay you can force it to always show the instance number instead.
- Renaming the TRACK display — right-click (Ctrl-click on Mac) or double-click the TRACK strip to give it your own name: type up to 4 characters (A-Z, 0-9, space), Enter confirms, Escape cancels. Leave it empty (or choose "Reset to DAW name" from the right-click menu) to return to the DAW track name. A name you set here wins over the DAW name and over "force instance number". It is saved with the session, not with presets.
- PRESETS button — opens the preset menu: factory presets plus your own preset files. See section 9.
- Status LED + label — green & "ACTIVE" when audio is processed, off when bypassed. During trial it shows "TRIAL Xd" or "TRIAL mm:ss" countdown.
- Version — current plug-in version, top right.
Link groups (LINK A / B / C)
The three LINK keys tie instances together across channels. Put every
backing-vocal instance in group A, and moving a fader on one
of them moves the same fader on all of them — relatively, like a
trim: each instance keeps its own offset, the group follows the gesture.
Buttons (CPU, LATENCY, BYPASS, DELTA) switch absolutely across the group.
An instance can be in several groups; linking works between instances in
the same DAW project.
Meters
- IN / OUT — input and output signal level, shown as
RMS (average energy over the block), not momentary peak.
RMS reflects the average level of the signal rather than its loudest
individual samples.
- VOICE — the detected voice probability, 0–100 %. This
is a confidence indicator, not a level — so its segments stay
green across the whole range (no yellow/red), even at
high percentages. A high reading just means "strong voice detected here."
RMS vs. peak. The IN/OUT meters show RMS (average) level. If
your DAW's channel meter is a peak meter, the two scales read on different
references by nature — use IN/OUT to watch the relative change (does OUT
track IN?) rather than as an absolute peak reference.
Processing engine
All processing in this version runs on the CPU — no GPU is
used or required, and there is nothing to configure. The AI model loads
in the background the first time audio flows, so inserting instances and
plug-in scans stay instant.
6. The Two Faders
STRENGTH (0 … 10)
The master amount of denoise / cleanup processing. Fader at 0
means the plug-in is effectively bypassed (gain = 1 across all
bands). Fader at 10 means full processing.
Use this fader to dial in how aggressive Clear Voice Live is. The
colour scale next to the fader in the GUI overview shows the working ranges:
3 to 5 is the normal range (at 3 a typical setup gets a
gentle denoise and soft feedback prevention; if you need more, move towards
5), 5 to 8 is a strong intervention that becomes audible,
and 8 to 10 is for emergencies only. Below 3 the plug-in
does very little.
DEREVERB (0 … 10)
Dries up the room: reduces the reverb tail that follows each syllable and
the reflections between words. New in 1.1, this fader drives two
engines at once — a continuous tail suppressor, plus a continuous
deepening of the AI model's own dereverb behaviour. At 0
the model runs in its neutral, reference-tuned state; raising the fader
pushes it smoothly towards maximum dryness. At 10 the
voice sounds tight and close-mic'd even in a genuinely reverberant room
(at the very top a slight thinning of the voice is normal — that is the
character of maximum dereverb, back it off a touch if you hear it).
Sessions and presets from v1.0.0 that used the SMOOTH fader still load
correctly — the parameter lives on and keeps working, it simply no longer
has a fader (the new model made manual smoothing unnecessary).
Important — Dereverb only works in combination with Strength.
If Strength is at 0, Dereverb does nothing — the entire processing pipeline
is bypassed. To hear Dereverb's effect, raise Strength first.
Dereverb is feedback prevention, too — use it even when there's no
audible reverb. Every feedback loop runs through the room: the
reflections that come back into the microphone are what keeps the loop
alive. Dereverb takes exactly those reflections out of the signal, so a
small amount (1 – 3) helps against acoustic coupling: in our live tests the
plug-in decouples more reliably and holds a feedback-prone monitor setup with
more headroom when Dereverb is not at 0. We recommend always running with a
non-zero Dereverb in live situations, even on a dry stage.
7. CPU & Latency Modes
CPU: LOW & HIGH
The CPU keys select how much processing power an instance may spend:
| LOW | HIGH |
| CPU | Ultra-low — cheaper per instance than
v1.0.0's SOFT mode. Run it on every channel of the show. |
The expensive one in ZERO latency — several times the
cost of LOW: the full AI model at its finest resolution. In LOW latency it
costs the same as LOW. |
| Processing | Lightweight voice engine. | The complete
AI model (trained on speech and singing) across the full
frequency range. |
| Sound character | Clean and open. Excellent on
already-decent signals. | The finest resolution — the most
detailed denoise and the most transparent voice. |
| Best for | About 90 % of all applications.
Feedback-critical live setups (loud monitors, mic close to the PA),
large channel counts, conference tables, backing vocals, and most lead
vocals too — in ZERO latency this is the mode to start with and, in most
cases, to stay with. | The finest denoising — broadcast, streaming,
recording, and the few live channels where the last bit of detail matters
more than maximum feedback headroom. Recommended in LOW
latency, where it costs no more than LOW. |
Feedback-heavy stage? Choose LOW. In setups with a lot
of acoustic coupling — loud wedges, handheld mics walking in front of the
PA — LOW is the more stable, safer mode and holds the
loop down best. HIGH resolves the spectrum more finely
and is the better-sounding choice for denoising; pick it whenever
feedback is not your main enemy. Two tools for two jobs.
What the modes cost, relative to each other
Measured per instance in Reaper on a Windows laptop and on a MacBook with
v1.1.0. Absolute numbers say more about the machine than about the plug-in, so
what follows is the relationship between the modes — that part holds
everywhere.
| Mode | Cost per instance | Reported latency |
| LOW · ZERO | The cheapest mode — roughly a third less than v1.0.0's SOFT | 0 samples |
| LOW · LOW latency | Around half of v1.0.0's HARD | 127 samples |
| HIGH · ZERO | The expensive one — a multiple of v1.0.0's HARD | 0 samples |
| HIGH · LOW latency | Same as LOW in this latency mode, around half of v1.0.0's HARD | 127 samples |
Two things to read out of this table. LOW in ZERO latency became
cheaper than v1.0.0's SOFT mode (about a third less), so the mode
you will run on most channels costs less than it used to.
HIGH in ZERO latency became more expensive than v1.0.0's HARD
mode — that is the deliberate price of the finer 256-band
resolution that makes its denoising so clean.
LOW latency changes the CPU picture completely. There, both
CPU settings run the same full-band engine, so HIGH costs no more
than LOW. If a channel can live with 2.65 ms — FOH-only chains,
streaming, broadcast, recording — that is by far the cheapest way to get
HIGH's sound, and it is where we recommend HIGH. Our rule of thumb:
LOW in ZERO latency for about 90 % of applications; HIGH when you want that
last bit of finesse, preferably in LOW latency.
Latency: ZERO & LOW
| ZERO | LOW |
| Output latency | 0.0 ms — no PDC. |
2.65 ms (127 samples at 48 kHz),
automatically reported to the host as PDC. |
| Processing | Gain changes are applied with zero delay. |
A short look-ahead lets the treble range be processed with finer
resolution. |
| Best for | Live shows, in-ear monitoring — anything where
the artist hears themselves. | Streaming, broadcast, recording,
FOH-only chains — anywhere a few ms are invisible. |
For live monitoring stay on ZERO. LOW's extra
milliseconds are compensated by DAWs during playback, but in a live
monitoring path they add to the roundtrip the artist feels.
Two safety buttons in the centre column. Both are locked by default
to prevent accidental clicks during a live show. Each button has a small
amber lock strip below it.
How the lock works
- Click on the lock strip below a button → toggles between
Locked (amber) and Armed (dark).
- Click on the button itself → toggles the action, but only if the
button is unlocked.
- The locked state persists across DAW sessions.
BYPASS
When ON, the plug-in is in audio bypass — the dry input passes straight
through. Useful for rapid A/B comparison ("how much better is this with
the plug-in on?").
DELTA
When ON, the plug-in outputs only the difference between the
processed and the dry signal. You hear what the plug-in is removing —
noise, room tail, feedback resonances.
Delta is a "nice to have" inspection tool. Never set values while
in Delta mode. Delta lets you hear what's being removed, but it
is not a tuning aid — the spectral character of the residual is not
representative of how aggressive the processing is. Set values with the
plug-in in normal mode, by ear, and use Delta only as a curiosity.
⚠ Warning — Delta Output Can Cause Feedback
Only enable DELTA when the channel is muted at the desk,
or when monitoring through closed-back headphones in solo.
Never
switch Delta on while the microphone is open through a monitor wedge,
in-ear pack, or PA — the residual signal from the AI gating contains
unpredictable spectral peaks that can drive the system into instant
feedback or sound horrendously to the audience. Delta is a studio /
diagnostic tool, not a live tool.
9. Presets
The PRESETS button in the header opens the preset menu.
The factory presets come in two flavours — one submenu per latency mode,
so every preset loads with the right latency setting for its job:
- Zero Latency — for live shows and monitoring
(0.0 ms).
- Low Latency — the same starting points with the
finer-resolution low-latency engine, for streaming/broadcast/recording.
- User — your own preset files, loaded from
Documents\Clear Voice Live\Presets\ — presets you saved
with v1.0 appear here automatically.
The factory presets
| Preset | CPU | Strength | Dereverb | Idea |
| Good starting point | LOW | 3 | 6 |
A safe, transparent default for almost any vocal channel. |
| Like other tools | LOW | 3 | 10 |
Close to that other company's sound at 100%. |
| Soft Denoise | HIGH | 4 | 0 |
Pure, fine-resolution denoising; the room stays untouched. |
| Handheld in Knee-position | LOW | 9 | 5 |
For speakers who let the handheld drop to their side and trust it
to still pick them up — aggressive bleed and feedback control for
a mic that is far from the mouth. |
| Lifesaver | LOW | 10 | 8 |
Everything the plug-in has. For situations that are already on
fire. |
| Make it sound good | HIGH | 5 | 10 |
The beauty preset: finest denoising plus a tight, dry voice. |
Saving and managing your own presets
- Save preset… stores the current settings (CPU mode,
Strength, Dereverb, plus the legacy parameters) as a
.cvlpreset file in the preset folder.
- Open presets folder opens that folder in
Explorer/Finder.
- The file name is the preset name. Rename a file to
rename the preset; copy files between machines to share presets. The
menu re-scans the folder every time you open it — files you drop in
appear immediately, no restart needed.
A .cvlpreset file is plain, readable text — you can view
and edit it in any text editor:
# Clear Voice Live Preset
version=3
engine_type=0 (0 = LOW CPU, 1 = HIGH CPU)
latency_mode=0 (0 = ZERO, 1 = LOW)
strength=100 (0–100, the fader shows this ÷10)
dereverb_amount=0 (0–100, the fader shows this ÷10)
Preset files written by v1.0 (without the latency_mode line)
load fine and default to ZERO latency.
10. Status, Trial & License
The header shows one of these states at all times:
| Header text | Meaning |
| ● ACTIVE | Licensed, audio is being processed. |
| ● BYPASS | You pressed the BYPASS button. Click again to engage. |
| ● TRIAL Xd | Trial mode, X days remaining. |
| ● TRIAL mm:ss | Trial mode, less than an hour remaining. |
| ● EXPIRED | Trial period has ended. Audio is bypassed. |
| ● TAMPERED | Trial integrity check failed. Audio is bypassed. Contact support. |
Trial behaviour
A 10-day fully-functional trial begins on first plug-in instantiation. The
trial is bound to the machine you started it on. Renewing the trial
requires support assistance. Version 1.1.6 starts a fresh 10-day
trial on every machine — also where an earlier 1.1.x trial has
already expired — so you can re-evaluate the update with your own setup.
When the trial expires, the plug-in keeps loading and shows an "Expired"
overlay, but audio is bypassed cleanly until a license is activated.
Activation
After purchase you receive a personal license file by
e-mail — it carries your name and is cryptographically signed. Loading it
is the whole "activation":
- Open the plug-in and click the title plate (About overlay) →
Load License…
- Select the license file (or paste its text). Done — the header shows
"ACTIVE" and the plug-in is licensed permanently.
There is no machine binding, no challenge/response and no internet
connection required — ever. The same file works on your studio
machine, your laptop and a rented FOH computer on show day (one license per person, usable on all your own machines, one at a time; see the EULA). New computer? Just load
the same file again. Keep it with your backups.
11. Tips & Best Practices
- Close the plug-in window during the show if you don't need
it. The GUI (meters, animations) costs a little extra CPU;
with the editor closed every instance runs in its leanest audio-only
path. Keeping a window open to watch the meters is fine too.
- Start with LOW in ZERO latency at Strength 0 and increase.
For about 90 % of applications that is the final setting. If a channel
deserves the last bit of finesse, switch it to HIGH — preferably in LOW
latency, where HIGH costs the same as LOW; in ZERO latency HIGH costs
several times what LOW does.
- If soundcheck time permits, try both CPU modes per channel.
Different voices respond differently to LOW vs HIGH. There's no
universal best choice.
- Feedback-critical monitor setup? Start on LOW. LOW is
the most stable mode against acoustic coupling; switch to HIGH where
sound quality is the priority and feedback headroom is comfortable.
- Set the order: Strength → Dereverb. Dial in the
noise reduction first, the room tail second.
- Run a small Dereverb even on dry stages — it is feedback
prevention. 1 – 3 on the Dereverb fader takes the room
reflections out of the loop and helps the plug-in decouple a
feedback-prone monitor setup; it costs nothing in sound.
- Lock the buttons before the show. Both Bypass and
Delta default to locked. If you've armed them during sound check,
lock them again before doors open.
- Never enable Delta on a live mic into a monitor or PA.
See section 8 — Delta can cause feedback. Use it muted or in
headphones only.
- One instance per channel. Don't chain multiple
instances of Clear Voice Live in series; the AI gating doesn't
compound usefully and you'll spend CPU twice.
- Insert first on the vocal channel, before compressors and
EQs. The plug-in expects a relatively clean dynamic input.
Heavy compression before the plug-in muddies the noise floor
and makes the AI work harder. In particular, do not place it
on a subgroup behind a compressor — the compressor constantly
lifts the noise floor between phrases, which the AI then chases and
you hear as pumping. Corrective EQ boosts (e.g. a low shelf) belong
before the plug-in, sound-shaping EQ after it.
- Close-mic sources only. Clear Voice Live is built for
one voice on its own microphone (handheld, headset, lectern, lavalier).
It is not suited for ambient, room or audience microphones —
by design, everything that is not the voice gets pushed down, which on
a room mic is the very content you want to keep.
- Conferences & panel discussions: one instance per
microphone always gives the highest quality — each channel gets its own
AI. But a very good option for large panels: route all microphones
through an auto-mixer (e.g. a Dugan-style mixer) into a single
group, and place one instance of Clear Voice Live on the
group. You save CPU, get consistent treatment across the
table, and the auto-mixer keeps unused mics out of the path before
they reach the plug-in.
12. Troubleshooting
The host doesn't show the plug-in at all
- Use the installer instead of copying files by hand. It
puts the bundle in the right place and installs the required Microsoft
runtime.
- If you must copy manually: copy the entire
Clear Voice Live 1.1.vst3 folder to
C:\Program Files\Common Files\VST3\ — that is the standard
VST3 location every host scans. (On macOS the equivalents are
/Library/Audio/Plug-Ins/VST3/ and
/Library/Audio/Plug-Ins/Components/.)
(...\Common Files\VST without the "3" is the old VST2
folder; most hosts also ignore
%localappdata%-based VST3 paths.)
- v1.1 has no runtime dependencies — it is statically linked and
self-contained. If you are still on v1.0 (or the host scanner
blacklists the plug-in on Windows 10 LTSC systems such as the Waves
LiveBox), install the Microsoft VC++ runtime once from
aka.ms/vs/17/release/vc_redist.x64.exe
(official Microsoft link), reboot, then Clear Caches and Rescan in the
host. The Clear Voice Live installer also installs it automatically.
- Pro Tools: shows no VST3 plug-ins at all — it loads AAX only.
Use a wrapper (Blue Cat PatchWork, DDMF Metaplugin) to run Clear Voice Live there.
- Apple Silicon / SuperRack Performer: on M-series Macs, Performer
lists only ARM-native third-party VST3 and only from its current version with
third-party VST3 support enabled. Run the host natively (not under Rosetta) and
check that third-party VST3 support is switched on.
- Make sure your host is 64-bit and actually scans VST3 plug-ins.
The plug-in is bypassed and shows "UNSUPPORTED SAMPLE RATE"
Your DAW or interface is running at a sample rate other than 48 / 96 / 192
kHz. Switch your project to one of the supported rates and reload the
plug-in. Most live setups use 48 or 96 kHz.
The plug-in is bypassed and shows "TRIAL EXPIRED"
Your 10-day trial has ended. Your audio passes through unprocessed
(clean bypass — nothing is muted). Purchase a license at
clearvoice.live and follow the
activation steps in section 10.
The plug-in is bypassed and shows "TAMPERED"
The trial integrity check failed. Contact
info@clearvoice.live for a reset
and include the machine fingerprint visible in the About overlay.
I changed my machine / restored a system image — is my trial or license affected?
Your license is never machine-bound: the same license file
activates on any computer you own. The trial re-binds itself
automatically since v1.1 when it detects that the machine has changed (new
system image, hardware swap) — the earlier
“trial file does not match” message from v1.0 no longer
occurs. If you ever see trial issues after a system change, write to
info@clearvoice.live.
My host lists two Clear Voice Live entries — which one is which?
If you owned v1.0.0 before, that is expected and intended. The entry named
Clear Voice Live 1.1 is the new version; the one named
Clear Voice Live is your original v1.0.0. They are separate
plug-ins for the host, so both can be inserted in the same session — useful
for an A/B on one channel. The About overlay of each instance shows its exact
version number if you ever need to be certain.
Updating from v1.0.0 — the two versions live side by side
v1.1.0 installs as its own product. It appears in your host
as Clear Voice Live 1.1, right next to your existing
Clear Voice Live v1.0.0, which is left completely untouched.
Nothing is replaced, nothing is overwritten, and uninstalling either one does
not affect the other.
- Existing sessions keep loading v1.0.0. A show file saved
with v1.0.0 opens with the v1.0.0 plug-in — v1.1 never takes over on its
own. To move a channel to the new engine, insert
Clear Voice Live 1.1 and remove the old instance.
- Your presets and your license carry over. They live
outside the plug-in bundle in
Documents\Clear Voice Live\,
and both versions read the same folder. Your own presets show up in
v1.1's User menu with nothing to import. Your existing license
unlocks v1.1 as well.
- Going back is trivial — the old version never left. Swap
the insert back to Clear Voice Live and you have the v1.0.0
sound again, mid-show if you have to. There is nothing to reinstall.
- Re-check your gain staging when you swap a channel over.
v1.1 no longer pulls the overall level down the way v1.0.0 did, so the
same fader positions give you a hotter, more present signal.
- Still on the trial? v1.1 starts its own fresh 10-day
trial, independent of where your v1.0.0 trial stands — evaluate the new
engine properly even if the old window already ran out. v1.1.6 starts
another fresh 10-day trial, even where an earlier v1.1 trial has expired.
(An installed license unlocks all versions, of course.)
I hear crackles / dropouts
Check your CPU meter first — there are two very different causes.
1) CPU is high / at the limit. Raising the buffer size only
hides this and defeats a zero-latency live plug-in — instead, free up CPU.
Open the Windows Task Manager (Ctrl+Shift+Esc) or macOS Activity Monitor and
close what competes for it:
- Background syncing (cloud-storage clients, OneDrive, Dropbox, iCloud)
- Other audio software running in parallel
- Antivirus or backup scans
- Browser tabs with heavy JavaScript / streaming
- Windows Update or system maintenance tasks
Close anything you don't strictly need during the show. If CPU is still
borderline, switch some HIGH-mode channels to LOW.
2) CPU is low (a few percent) but you still hear crackles.
Then it isn't processing power — it's real-time delivery: an audio,
graphics or USB driver missing its deadline ("DPC latency"). The plug-in's own
processing is glitch-free here (confirm with an offline render — it will be
clean). Work through these in order:
- Power plan → High Performance (Windows) / disable
low-power & App Nap modes (macOS). Laptops throttle the CPU on
balanced plans — a common source of timing glitches.
- Check your drivers — a misbehaving background driver
(graphics, Wi-Fi, USB) is the usual culprit. Updating your graphics and
audio-interface drivers, and disabling Wi-Fi power-saving, clears most of
these crackles.
The voice sounds nasal / hollow
Strength is too high. Reduce it until the voice character returns. If
the noise floor still bleeds through at lower Strength, try switching
the CPU mode from LOW to HIGH and re-dialling Strength — the full model
reaches deeper at gentler settings.
I can't hear what Dereverb does
Dereverb requires Strength to be non-zero — it's part of the processing
pipeline that Strength enables. Set Strength first, then bring up
Dereverb.
13. Support
Questions, bug reports, license issues:
Email: info@clearvoice.live
Web: clearvoice.live
Open-source licenses for components used in this product are listed inside
the plug-in's About overlay and at
clearvoice.live/open-source.html.
End-User License Agreement: clearvoice.live/eula.html.
Privacy Policy: clearvoice.live/privacy.html.