feat: 0.5.0 Capture release — dictation, MCP, personalities (#544)

* feat(capture): dictation, personalities, 0.5.0

Ships the Capture release end to end. Global-hotkey dictation with
synthetic paste into the focused app on macOS and Windows, an on-screen
pill across recording / transcribing / refining, customizable push-to-
talk and toggle chords, and an accessibility-permission prompt scoped to
Settings → Captures with inline re-check feedback.

Voice profiles gain optional personalities that power compose / rewrite /
respond actions via a local Qwen3 LLM — shared with refinement, so there
is one local LLM in the app, not two.

Refinement hardened with deterministic Whisper-loop collapse before the
LLM sees the transcript, per-capture flag snapshots for re-runs, and a
ten-transcript evaluation harness across every bundled refinement size.

Version bump 0.4.5 → 0.5.0.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* feat(mcp): local MCP server exposes voicebox.* tools to AI agents

Mounts FastMCP at /mcp (Streamable HTTP) so Claude Code, Cursor,
Windsurf, and the VS Code MCP extensions can call voicebox.speak,
voicebox.transcribe, voicebox.list_captures, and voicebox.list_profiles
against the running Voicebox server.

Backend
- new backend/mcp_server package (tools, middleware, profile resolve,
  pub/sub events); named mcp_server to avoid shadowing the installed mcp
  PyPI package FastMCP imports internally
- app.py migrated from @app.on_event to lifespan= so FastMCP's session
  manager cohabits with Voicebox's startup/shutdown
- new MCPClientBinding table + /mcp/bindings CRUD; ClientIdMiddleware
  reads X-Voicebox-Client-Id into a ContextVar and stamps last_seen_at
- profile resolution precedence: explicit -> per-client binding ->
  capture_settings.default_playback_voice_id
- POST /speak REST wrapper for non-MCP callers (shell, ACP, A2A)
- GET /events/speak SSE broadcasts speak-start / speak-end so the pill
  surfaces agent-initiated speech
- backend/mcp_shim proxy (plain httpx) for stdio-only MCP clients
- PyInstaller spec updates + new --shim build target (~18 MB)

Frontend
- Settings -> MCP page with HTTP / stdio / claude-mcp-add copy snippets,
  default voice picker, per-client bindings table, connection status
- useMCPBindings, useSpeakEvents hooks
- CapturePill gains 'speaking' state; DictateWindow subscribes to SSE
  and emits dictate:show so the Rust side surfaces the pill window

Native
- tauri.conf.json externalBin now includes voicebox-mcp
- show_dictate_window helper + dictate:show listener in main.rs
- (also in this commit: InputMonitoringGate UX, hotkey_monitor tweaks,
  landing footer/navbar updates, new overview docs for captures /
  dictation / mcp-server / voice-personalities)

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* feat(mcp): Rust-owned speaking pill with self-contained audio playback

The pill window now surfaces for agent-initiated speech without main-window
involvement. Rust subscribes to /events/speak via a tokio task + reqwest
streaming body (speak_monitor.rs), shows the pill, and forwards events to
the dictate webview over Tauri's event bus. The pill plays audio via a
plain HTMLAudioElement and emits dictate:hide when playback ends. The
pill stays hidden through the ~1 s generation wait and only surfaces when
audio actually starts, with the counter armed at that moment.

Fixes a shared-dict mutation in mcp_server/events.publish() that caused
the second subscriber (Rust speak_monitor) to receive `event: message`
instead of named speak-start/speak-end frames. Also teaches the speak_monitor
parser to handle CRLF framing (sse-starlette default). Main-window
AudioPlayer now skips autoplay for source in {mcp, rest} to avoid
double-play when both windows are alive.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* readme and dev script

* feat(capture): gate global hotkey on dictation readiness checklist

Stops the "stuck pill" failure where pressing the chord with missing
STT/LLM models triggers a recording that has nowhere to land. The
hotkey now stays disarmed until every gate (models downloaded, Input
Monitoring + Accessibility granted) is green; the empty-state checklist
in CapturesTab surfaces each unmet gate with a one-click action and
auto-arms the chord once everything turns green.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* color

* model download status

* progress

* personality: bool API, i18n across the app

- Collapse intent tri-state (respond/rewrite/compose) to `personality: bool` on /generate, /speak, and voicebox.speak. Drop respond entirely; keep compose as a standalone button via /profiles/{id}/compose. Remove /rewrite, /respond, and /speak profile endpoints.
- FloatingGenerateBox: Wand2 persona toggle + Dices compose button appear when the selected profile has a personality. ProfileCard badges Wand2 alongside the effects Sparkles.
- MCP bindings: default_intent column → default_personality: bool. Migration drops the legacy column.
- i18n: en / ja / zh-CN / zh-TW translation files filled out and wired through the capture, server, and profile UI.

```ts
voicebox.speak({
  text: "Deploy complete.",
  profile: "Morgan",
  personality: true, // rewrite through the profile's personality LLM
});
```

* i18n: GenerationPage sidebar copy

* fix: BOOL import for windows crate 0.62

BOOL moved from Win32::Foundation to windows::core in 0.62.

Co-Authored-By: Claude Opus 4.6 (1M context) <[email protected]>

* fix(capture): layout-aware V keycode for synthetic paste on macOS

macOS apps match Cmd+V against the layout-translated character via NSMenu
key equivalents, so posting kVK_ANSI_V (= 9, the QWERTY V position) on
Dvorak produces Cmd+. and never triggers Paste. New keyboard_layout
module resolves the active layout's V keycode via
TISCopyCurrentKeyboardLayoutInputSource + UCKeyTranslate, caches it in an
AtomicU16, and refreshes on kTISNotifySelectedKeyboardInputSourceChanged.
All TIS calls run on the main thread (init from Tauri setup; observer
callback delivered to the main runloop); synthetic_keys::send_paste
reads the cached value once per paste. Falls back to kVK_ANSI_V when
resolution fails or the active input source carries no Unicode key
layout data.

Windows is intentionally left on hardcoded VK_V — SendInput delivers
WM_KEYDOWN with wParam = VK_V to the target regardless of the active
layout, which is why `Send "^v"` works for AutoHotkey on Dvorak Windows.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(capture): cooperative app activation for synthetic paste on macOS 14+

macOS 14 deprecated NSRunningApplication.activateWithOptions: in favour
of a cooperative-activation pattern: the caller first yields activation
rights to the target, then the target activate()s against the tightened
Sonoma foreground rules. Without the yield, activate() on 14+ sometimes
silently fails or only bounces the dock icon — the exact "paste lands in
the wrong app" symptom we were previously one API break away from.

activate_pid now discovers the 14+ selector via respondsToSelector: and
branches: on 14+ it yieldActivationToApplication:'s from
NSRunningApplication.current then calls -activate on the target; on
11–13 it stays on -activateWithOptions: (still the only option). Both
branches propagate the BOOL return — if activation is refused we error
out before clobbering the clipboard instead of silently proceeding.

The respondsToSelector: result is cached in a OnceLock so the probe
isn't repeated on every paste.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(capture): conditional clipboard restore + always-attempt on paste failure

Two bugs in paste_final_text' clipboard handling:

1. Restore was unconditional. If the user ⌘C'd in the target app during
   the 400 ms paste-consume window — or a clipboard history tool (Paste,
   Pastebot, Maccy) or Universal Clipboard sync snapshotted our staged
   text — the blind restore overwrote their newer content with the
   pre-paste snapshot, silently losing user data.

2. send_paste' errors were propagated with ? before the restore, so a
   CGEventPost / SendInput failure left the user's clipboard stuck on
   the transcript.

Fix folds both into one pattern: capture the post-write change count,
re-read it after paste-consume, restore only when they match (plus treat
a change-count read failure as "unknown, don't overwrite"). Isolate
send_paste's error so the restore runs regardless of paste success, then
propagate the paste error after.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* chore(deps): pin rdev to jamiepine/rdev fork

Upstream Narsil/rdev has shipped no release since 2023-06 (crates.io
still serves 0.5.3), so the Sonoma main-thread fix we depend on — PR
#147, applied at hotkey_monitor.rs:184 — is only reachable via a git
pin. A pin to a third-party repo breaks the build whenever the remote
force-pushes, renames, or is taken down, and Cargo does not durably
cache git-dep archives the way it does crates.io tarballs.

Forking to jamiepine/rdev at the same SHA removes that failure mode
without changing crate behavior and gives us a place to cherry-pick
future OS-compatibility fixes on our own timeline. The SHA was verified
to exist on the fork before re-pinning.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(mcp): idle timeout + escalating backoff for speak-SSE monitor

Two reliability gaps in the /events/speak subscriber:

1. resp.chunk().await had no idle timeout. A backend that accepts the
   TCP connection but stops producing frames (deadlocked SSE endpoint,
   zombie process) would block the task forever without reconnecting.
   The pill window would never surface for agent-initiated speech and
   there would be nothing to log. Backend emits a `:ping` heartbeat
   every 15 s, so 45 s without any data is now treated as a dead
   stream — the task errors out and the reconnect loop takes over.

2. Flat 2 s backoff escalates nowhere. Logs fill with reconnect lines
   when the backend is down for minutes, and a backend that accepts +
   immediately closes connections (no data) spins the loop tightly.
   Backoff now escalates 500 ms → 30 s on unproductive rounds and
   resets only when at least one frame arrives (the connection was
   genuinely productive, not just accepted).

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(refinement): character-level loop collapse + pytest coverage

The word-level pass catches single-word Whisper loops ("URL URL URL…")
but misses two common hallucination patterns the PR had to claim as
"edge cases":

1. Multi-word English loops — "thanks for watching thanks for watching…"
   × 6 sails through because no two consecutive tokens are identical
   after text.split().
2. CJK loops — "謝謝觀看" × 7 sails through because text.split() returns
   a single unsplit token for the whole loop (no whitespace between
   characters).

Add a character-level second pass: a non-greedy regex finds any 2–60
char substring that repeats min_run+ times immediately after itself and
strips the run. The 2-char floor keeps emphasised single-letter runs
("wooooooow") intact. The 60-char ceiling covers every observed
Whisper tail hallucination ("Please like and subscribe to my
channel.", "Subtitles by the Amara.org community") while staying short
enough that coincidental long-phrase repetition in legitimate speech
doesn't hit the threshold. Whitespace normalisation only runs when the
pass actually stripped something, so untouched transcripts keep their
original spacing.

New test_refinement_collapse.py gives the pre-processor its first
deterministic unit-test coverage: 17 tests pinning the word-level
legacy behaviour plus the new multi-word English / CJK / Japanese /
emphasis-preservation cases.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(db): graceful fallback when SQLite < 3.35 on MCP bindings migration

SQLite gained ALTER TABLE … DROP COLUMN in 3.35 (Mar 2021). Production
PyInstaller builds bundle Python 3.12 which links to SQLite 3.40+ so
that path is always safe, but a dev running the backend directly on
Ubuntu 20.04 (3.31) or Debian 11 (3.34) would crash on first startup
trying to drop the legacy default_intent column.

Add _supports_drop_column(engine) — returns True on non-SQLite
dialects (Postgres / MySQL have supported DROP COLUMN for decades) and
gates on the runtime sqlite_version for SQLite. When unsupported, log a
warning and leave the unused column in place: SQLAlchemy only maps
declared columns, so a stray default_intent column does no reads or
writes and can't interfere with runtime behaviour.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(mcp): correct lifespan shutdown order — drain MCP before unloading models

The inline lifespan ran _run_shutdown inside the MCP context, so the
TTS / Whisper / LLM models were unloaded *before* FastMCP's __aexit__
got a chance to cancel its in-flight session tasks. Any MCP request
mid-generate at shutdown time would crash on "model unloaded" instead
of receiving a clean session-cancelled error.

Rewire via compose_lifespan (which was already defined in
mcp_server.server for exactly this purpose but never used):
AsyncExitStack enters factories in order and exits in LIFO, so
MCP teardown fires first — cancelling sessions — and _run_shutdown
runs after nothing is holding the models. Smoke test shows the
log order flipped as expected:

  Ready
  StreamableHTTP session manager started
  ... running ...
  StreamableHTTP session manager shutting down   ← was last, now first
  Voicebox server shutting down...               ← was first, now last

As a side benefit, _run_shutdown is now paired with _run_startup via
try/finally inside voicebox_lifespan, so a partial startup (models
half-loaded, MCP __aenter__ fails) still unloads whatever was loaded
instead of leaking it to process exit.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(mcp): stamp last_seen_at on /speak too + tighten path predicate

POST /speak is a REST wrapper around voicebox.speak for agents that
don't talk MCP (shell scripts, ACP, A2A). It reads X-Voicebox-Client-Id
and uses it for the same per-client profile resolution + default
personality lookup the MCP tool does (speak.py:39-64), so its callers
are first-class clients — but the ClientIdMiddleware only stamped
last_seen_at on /mcp* paths. REST speak callers showed up as "never
seen" in Settings → MCP despite actively acting on their bindings.

Widen the stamp predicate to an explicit ("/mcp", "/speak") prefix
list, and require a path boundary on match so future routes named
/mcpfoo or /speakers don't silently inherit the stamp via the prefix.
New test_client_id_middleware.py pins the scope with 17 parametrised
cases (both the allowed set and the overlap cases that must not match).

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* feat(captures): scrubbable WaveSurfer player for capture detail view

Replace the placeholder fake-waveform + play button in CapturesTab's
audio card with a real CaptureInlinePlayer (wavesurfer.js). The player
renders the actual waveform, lets users scrub through the clip, and
shows a proper current/total timestamp pair in place of the
duration-only label.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* feat(ui): persist selectedProfileId across sessions

Wrap useUIStore in zustand/middleware's persist under the key
voicebox-ui. partialize only selectedProfileId so volatile UI state
(dialog open flags, form drafts, engine/voice pickers, sidebar) stays
in-memory as before — but reopening the app no longer loses whichever
profile the user was last working with.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* feat(captures): mirror readiness checklist into the settings sidebar

The six-gate checklist only rendered in the CapturesTab empty state, so
a user already on the settings page had no single surface showing which
gate was red — the inline InputMonitoringNotice covered one, the model
pickers covered another, and Accessibility was only hinted at by the
auto-paste toggle. Mirror the same component into the right sidebar of
the settings page so every gate (STT model, LLM model, Input Monitoring,
Accessibility, plus the hotkey toggle in the main column) is always
visible while the user configures dictation.

New compact prop on DictationReadinessChecklist drops the centered
header and empty-state max-width so it fits the 280 px sidebar next to
the existing About / Differences blocks. Callers in compact mode own
the heading — CapturesPage reuses the existing captures.readiness.title
key (present in en / ja / zh-CN / zh-TW already) as an h3 matching the
sibling sidebar sections.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* feat(captures): move sidebar checklist below differences + hide when all green

Two small follow-ups to the sidebar checklist placement. Move it below
the What's different section so the sticky top of the sidebar stays the
page's narrative context (About → differences) and the checklist reads
as a status panel rather than preamble. Gate the whole block on
!readiness.allReady so once every gate is green the sidebar drops back
to just About + What's different — no value in real estate full of
checkmarks.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(captures): refetch readiness immediately after STT/LLM model swap

useCaptureSettings updated its own cache optimistically but never
invalidated ['capture-readiness'], so for up to 5 s (the poll interval)
after switching stt_model or llm_model the checklist kept showing the
previous model's ready/missing state. The backend endpoint resolves
the model live on each call — it was just the frontend cache that
lagged. Invalidate in onSettled only when the patch touched a model
field, so unrelated updates (chord keys, toggles) don't pay for a
refetch.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* fix(captures): hide macOS-only copy when running on Windows / Linux

Two surfaces leaked macOS-specific copy onto other platforms:

1. The Input Monitoring + Accessibility rows in the readiness
   checklist rendered everywhere. On Windows/Linux the Rust permission
   stubs return true, so the rows showed as permanent green checkmarks
   with copy like "macOS allows Voicebox to detect your global
   shortcut." — nonsense when you're on Windows. Gate both rows on a
   userAgent-based isMacOS check so they only render where the
   underlying TCC permission actually exists.

2. The global-shortcut setting description ended with "macOS will ask
   for Input Monitoring permission the first time you turn this on."
   That sentence rendered on every platform. The readiness checklist
   already surfaces the TCC requirement at the right moment on macOS,
   so the description doesn't need the platform note — drop it from
   en / ja / zh-CN / zh-TW.

Other macOS strings (AccessibilityNotice, InputMonitoringNotice, their
"stillMissing" hints) are already gated behind the Rust permission
booleans returning false, which never happens on Windows/Linux, so they
stay inert without further changes.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* feat(capture): swap the rdev fork for keytap 0.2, delete local chord state machine

Dep swap:
- Drop the git-pinned jamiepine/rdev fork we were carrying since the
  upstream crate is abandoned.
- Depend on keytap 0.2 from crates.io — our own cross-platform global
  keyboard tap crate. Clean shutdown via Drop, Sonoma-safe by design
  (no TSMGetInputSourceProperty calls off the main thread, so
  `set_is_main_thread(false)` is gone), and properly versioned.

Chord engine rewrite:
- Delete hotkey_monitor.rs's internal Chord state machine (Match enum,
  KeyEvent enum, step()/classify() methods, associated unit tests).
  keytap's ChordMatcher subsumes it: Momentary chord for PTT,
  add_toggle() for Toggle-to-talk, longest-match resolution, sticky-end
  for Toggle. Net: -80 LOC in hotkey_monitor.rs; the remaining module
  is the dispatcher loop + Effect→Tauri translation.
- Preserve the PTT→Toggle "RestartRecording" upgrade signal. keytap
  emits End(PTT)+Start(Toggle) atomically (same Instant) when the held
  set upgrades from a shorter chord to a longer superset. The
  dispatcher peeks at the matcher with a 5 ms recv_timeout after any
  End and coalesces the pair into Effect::RestartRecording so the
  frontend still gets the "discard the transition-moment audio" signal
  instead of an unrelated Stop+Start pair.
- HotkeyMonitor::update_bindings now actually tears down the tap on
  empty bindings instead of leaving an idle CGEventTap around. New
  bindings rebuild the matcher and the dispatcher thread from scratch.

key_codes.rs:
- Rewrite the browser-code → Key table against keytap's cleaner Key
  variant names (`A`..`Z` not `KeyA`..`KeyZ`, `Digit0`..`Digit9` not
  `Num0`..`Num9`, `ArrowUp` not `UpArrow`, `AltLeft`/`AltRight` instead
  of `Alt`/`AltGr`, `Period` not `Dot`, …). On-disk chord string
  format (W3C `KeyboardEvent.code` identifiers) is unchanged, so
  capture_settings rows written before the swap round-trip identically.
  Legacy aliases (`Alt`, `AltGr`, `Num0`, `UpArrow`, `Dot`, …) kept for
  forward-compat on old rows.

main.rs / input_monitoring.rs:
- Update the few doc comments that referenced `rdev::listen` to
  describe keytap's Tap; no behavioural change.
- build_chord_bindings now imports from keytap::Key.
- enable_hotkey / disable_hotkey / update_chord_bindings reach into
  HotkeyMonitor via &mut since apply()/update_bindings() now mutate.

Tests live in keytap now (22 chord-related tests in keytap 0.2,
including the PTT→Toggle upgrade scenario that used to be tested in
hotkey_monitor.rs). Voicebox's hotkey_monitor.rs is thin enough that
local testing would be trivia.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

* chore(deps): bump keytap 0.2 → 0.4 for macOS modifier-events fix

0.2 read CGEventFlags via CGEventGetIntegerValueField(event, 0x81),
which is not a valid CGEventField id — macOS silently returned 0, so
FlagsChanged events produced no KeyDown / KeyUp for any modifier key
and the PTT / toggle chords never armed on macOS. 0.4 uses the
documented CGEventGetFlags(event) API.

0.3 (tracing / serde / Fn / IntlBackslash) is picked up as a free
consequence; no API surface we depend on changed.

* perf(captures): stop polling readiness once both models are green

useQuery was firing GET /capture/readiness every 5s forever, and also
on every window focus. Once stt.ready and llm.ready are both true the
answer can only change when the user swaps a model in settings, and
useSettings already invalidates the query on that path — the polling
was pure noise.

Gate both refetchInterval and refetchOnWindowFocus on "not fully ready"
so we fall silent once the checklist is green.

* feat(ui): theme settings, stories polish, track editor restructure

- add dark/light/system theme with persisted choice + OS change listener
- restyle stories sidebar (search, item layout, border) to match captures
- move floating generate box to right column of stories, add top fade mask
- story track editor: sticky track labels aligned via flex rows, custom scrollbar with left/right zoom handles
- capture pill light mode pass, fix inline waveform progress color
- pull mcp_server hidden imports into the pyinstaller spec
- notarization doc draft

* fix mlx llm bundling

* feat(ui): shared ListPane primitive + misc polish

ListPane is a compound component (Header / TitleRow / Title / Actions /
Search / Scroll) that owns the relative wrapper, faded right divider
(50px top fade), top scroll mask, and absolute-positioned header used by
every list-detail tab. Wires up CapturesTab, StoryList, and EffectsList.
EffectsTab gets -mx-8 / pr-8 to match the edge-to-edge layout used
elsewhere.

Other changes:
- MCPPage: native <select> → shadcn <Select> for default voice and
  per-binding voice pickers
- Button outline variant: add hover:border-accent
- Drop hover:text-destructive from trailing delete buttons
  (HistoryTable, GpuAcceleration, GpuPage, EffectsChainEditor,
  EffectsDetail)
- HistoryTable empty state moved behind t('history.empty')
- StoryContent scroll padding pt-14 → pt-16
- backend health reports the captures dir
- landing CapturesMockup: "Send to" → "Export" with Download icon
- CHANGELOG: drop [Unreleased] personality section

* fix(captures): Play As autoplay + default voice + orphan recovery

- Hand /generate ids to the global SSE watcher so playback fires on completion. The mutation onSuccess was checking audio_path on a queued row, which is always empty — autoplay never ran.
- Bind the Play As voice selection to capture_settings.default_playback_voice_id, kept in sync with the Settings → Captures and Settings → MCP pickers. Picking from the split-button dropdown writes back to settings.
- Extract AudioBars from HistoryTable into a shared component; use it for the Play As generating state in place of Loader2.
- Stop the active-state hover from flashing white text when the button is in its lighter accent/10 fill.
- Drop the gradient avatar swatches from the Settings → Captures voice dropdown.
- Backend: when the gen worker exits without writing a terminal status (e.g. SQLite lock racing the failed-status write inside its own exception handler), the cancel endpoint now flips the row to failed instead of 409-ing. Worker also force-fails on its way out as a belt-and-suspenders.

* fix(captures+chord): Stop button stops, ChordPicker accepts shorter chords

Two unrelated correctness bugs caught in PR review:

- The Play As "Stop" button was wired to handlePlayAs() unconditionally, so clicking it during playback kicked a fresh generation instead of halting. Now pauses the player when the click came from the main button while playbackState is 'playing'. Picking a different voice from the dropdown still kicks a new generation as before.
- ChordPicker tracked the peak set of held keys but seeded the peak from initialKeys, so a user who opened the picker with a 3-key chord saved couldn't replace it with a 2-key chord — the candidate length never beat the seed. The peak now resets on the first press of a fresh sequence (when no keys were held immediately prior), then grows monotonically within that hold.

* fix(settings): honor explicit null on nullable fields, ignore on the rest

Routes were calling model_dump(exclude_none=True), which drops every
client-sent null before it reaches the service. The service then layered
on its own `if value is not None` guard. Net effect: setting a nullable
column back to null was a no-op — the MCPPage default-voice picker sends
null when the user picks "no default" and the row was silently keeping
whatever was there before.

Switched the routes to exclude_unset=True so absent fields stay absent
but explicit nulls survive the dump, and centralised the per-field
nullability check in the service. The check inspects the SQLAlchemy
column metadata so non-nullable columns (stt_model, llm_model, the chord
key lists) still drop nulls instead of crashing the request, while
default_playback_voice_id can finally be cleared.

* fix(captures): clean up audio files when create_capture fails

The create flow wrote raw audio (and a transcoded .wav for non-wav
sources) to data/captures before the DB row was committed, so any
failure between the write and the commit — a webm that decoded to a
0-length array, a whisper model that errored mid-transcribe, a SQLite
contention on the commit — left the audio on disk with nothing pointing
at it. Over enough flaky uploads the directory grows without bound.

Now every path written before the commit is tracked in a list, and the
whole stretch from the first write to db.commit() runs inside a
try/except that unlinks each tracked file on raise and re-raises. The
transcode branch removes the raw file from the cleanup list only when
the unlink actually succeeds, so an OSError on the raw-path delete
still hands cleanup the original blob to retry.

* fix(mcp): restrict voicebox.transcribe(audio_path=...) to loopback

audio_path mode took any absolute filesystem path and returned its
decoded contents as transcribed text with no caller verification beyond
the existence/size checks. The X-Voicebox-Client-Id middleware records
the header but never rejects an absent or fake one, so a Voicebox bound
to 0.0.0.0 (the documented "remote access" mode) was effectively an
unauthenticated arbitrary-local-file read primitive.

The middleware now stashes the request's remote address in a ContextVar
alongside the existing client_id, and audio_path mode refuses anything
that doesn't parse as a loopback address (IPv4 127.0.0.0/8, IPv6 ::1).
audio_base64 mode is unchanged — that path was always bounded to bytes
the caller already has.

Loopback callers (the Tauri webview, local CLI scripts, MCP clients on
the same machine) keep working. Remote callers now have to send the
audio over the wire if they want it transcribed.

* fix: PR review nits — response shape, landing copy, form reset

- /llm/generate's "model is downloading" branch was raising HTTPException(202, detail={...}), which wraps the payload in {"detail": ...} and forces clients to parse a success status as if it were an error. Switched to JSONResponse so the payload sits at the top level.
- The landing page's "Language Models" card advertised "Qwen 3.5" with sizes 4B/2B/0.8B; we ship Qwen3 at 0.6B/1.7B/4B. Aligned to what's actually in the binary.
- ProfileForm's discard-draft button reset the form without touching `personality` or `avatarFile`, so stale persona text and an attached avatar would survive the discard. The other three resets in the file already include both fields — this brings the discard path in line.

* perf(mcp): move last_seen_at stamp off the request path

ClientIdMiddleware was running the SQLAlchemy SELECT/INSERT/UPDATE/COMMIT
inline on the event loop after every /mcp/* and /speak request. SQLite
serialises writes, so concurrent MCP traffic queued behind the stamp
write — the response sat waiting on a side-effect that the client never
needs in band, and SSE streams would stall briefly per request.

The middleware now hands the stamp to asyncio.to_thread via a fire-and-
forget create_task so the response returns immediately and the write
runs on the default executor. A module-level set keeps strong refs to
in-flight tasks (per asyncio docs) so the GC can't collect them mid-
write. The fallback path runs the stamp inline if no loop is available
(tests/oddball callers) rather than silently dropping it.

* fix(dictate): force-dismiss the speaking pill when SSE never comes back

The pill subscribed to /generation/{id}/status to know when to start
playback, but EventSource.onerror was a no-op — auto-reconnect was the
intended recovery for transient drops. The gap: if the backend deletes
the gen row mid-flight or the connection silently dies in a way the
browser keeps retrying without ever getting a status event, the pill
sits in 'speaking' forever and the user has no way to clear it.

Added a 60-second hard cap that arms when the SSE opens and clears the
moment any real status event lands. If it fires while the pill is still
on the same id and audio never started, it force-dismisses. Same idea
as the existing post-speak-end 15s grace, but covers the case where the
backend never says anything at all.

* fix: i18n cleanup + readiness checklist effect cadence + ChordPicker shadow

- DictationReadinessChecklist was constructing downloadByModel as a fresh Map every render and listing it in the cleanup effect's deps. With the 1 s polling cadence and arbitrary parent rerenders the effect ran more often than it needed to. Memoised the Map on activeTasks; the effect now keys off the memo's identity.
- zh-CN persona tooltipActive/ariaLabelActive matched their inactive twins byte-for-byte ("以人物设定朗读"). The other locales differentiate the active state with a -ing / -中 suffix; zh-CN now reads "正以人物设定朗读" when active.
- personalityPlaceholder was a ~290-character paragraph that doubled as both the example text and the explanation, repeating most of what personalityHint already said. Trimmed to the example only and folded the explanation + leave-blank consequence into the hint, across all four locales.
- Refinement model size keys were size06 / size17 / size4. Renamed the 4B variant to size40 so the decimal padding is consistent.
- ChordPicker's open-effect bound a window.setTimeout id to a local `t`, shadowing the i18n `t` from useTranslation. Renamed to timeoutId.

* perf(settings): persist generation sliders on release, not per pointer-move

Both sliders on the generation settings page were calling update() —
which is a React Query mutation that PATCHes /settings/generation —
inside onValueChange. Dragging the chunk-limit slider from 800 to 3000
fired a request per pointer-move pixel, and a mid-drag failure plus
optimistic rollback would leave persisted state visibly out of sync
with the thumb position.

Local state now mirrors each slider during a drag and the persist
happens once on Radix's onValueCommit (pointer-up / keyboard-release).
useEffects keep the local state in sync if the persisted value changes
out-of-band — another window editing the same setting still updates the
slider position cleanly.

* chore(backend): Ruff lint pass — deprecated APIs, exception leaks, dead patterns

Mechanical sweep of items called out in the PR review:

- qwen_llm_backend: AutoModelForCausalLM.from_pretrained(torch_dtype=…) is deprecated in transformers ≥4.41 in favor of dtype=. Renamed.
- routes/llm: try/except around backend.generate() raised HTTPException(500, detail=str(e)) which leaks stack traces / paths to clients and trips Ruff B904. Now logs the original exception server-side and hands the client a generic message; chained via `from e` to preserve traceback context.
- mcp_bindings + mcp_server/context: datetime.utcnow() is deprecated since 3.12. Switched the two assignment sites to datetime.now(timezone.utc). The schema-level `default=datetime.utcnow` defaults in database/models.py are left for a later schema-aware pass.
- routes/generations: `logger = …` sat between two import blocks (Ruff E402). Moved below imports.
- mcp_server/server + tests/test_refinement_samples: typing.Callable / typing.Iterable have been preferred-via collections.abc since 3.9 (Ruff UP035).
- routes/events: `except asyncio.TimeoutError` aliases plain `TimeoutError` since 3.11 (UP041).
- services/captures: hoisted WHISPER_NATIVE_FORMATS to module scope (was a function-local UPPER_SNAKE that tripped N806) and replaced the raw_path.unlink try/except OSError-pass with contextlib.suppress (SIM105). Semantic equivalence preserved — written_files.remove(raw_path) still only runs when unlink succeeds because it sits inside the suppressed block after the unlink call.
- database/migrations: hoisted the duplicate `import sqlite3` from inside two helper bodies to a single module-level import.

* feat(stories): regenerate action on clips and the chat list dropdown

The track editor's clip toolbar now has a regenerate icon next to Delete; clicking it kicks a fresh take of the selected clip's underlying generation through the same /generate/{id}/regenerate path the History table uses, and pushes the id into the global pending set so the SSE watcher picks it up. The chat list's per-item dropdown gets the same action between Play-from-here and Remove. Translation keys added under storyContent.itemActions / storyContent.toast across all four locales.

* feat(stories): import external audio into the timeline (drag-drop + picker)

You can now drop a music file onto the story content area or pick one through the new "Import audio" button in the add-clip popover. Both call POST /generate/import which writes the file to data/generations/<id>.<ext>, probes duration via librosa, and inserts a Generation row pointing at a singleton "Imported Audio" profile (created lazily on first import). The existing addStoryItem flow takes over from there — the timeline doesn't care that the row didn't come out of TTS.

Engine field on the row is "import"; it's surfaced on StoryItemDetail so the chat list shows a music icon instead of the (missing) profile avatar and both the dropdown and the track-editor toolbar hide the Regenerate action — there's nothing to regenerate. Accepted formats: wav/mp3/flac/ogg/m4a/aac/webm, capped at 200 MB. Translation keys added across en/ja/zh-CN/zh-TW.

* fix(audio): serve real Content-Type so imports decode in WaveSurfer

/audio/{id} and /audio/version/{id} hardcoded media_type="audio/wav" on
the FileResponse. That was a no-op when every generation came out of
TTS (everything on disk was a .wav anyway), but imported audio keeps
its source format — .mp3 / .m4a / .ogg — and the WaveSurfer MediaElement
backend uses an <audio> tag that checks Content-Type before letting the
clip play, so an MP3 announced as audio/wav silently failed to load.

Both endpoints now derive the type via mimetypes.guess_type and fall
back to audio/wav for unknown suffixes. Download filenames also keep
the real extension instead of always saying ".wav".

* feat(stories): zoom bar bounds tracked to project length, default 60s scope

The track editor's zoom was clamped to a hardcoded [10, 200] pixels-per-second range, which had no relationship to the project — on a 4-minute story a "max zoom out" of 200 px/s still required scrolling, and on a 5-second story you could zoom all the way in to where every clip was a tiny sliver. Reframed the bounds in the unit the user actually thinks in: how many seconds of timeline are visible at once. Min scope is 10 s (most zoomed in), max scope is the entire project, and the default lands on a 60 s scope (or the full project, whichever is shorter) once the editor measures its visible track width on first mount.

The pixels-per-second value still lives in component state (because every downstream calculation already uses it) but minPps/maxPps are computed from `containerWidth − LABEL_COL_WIDTH` and the project's effective duration, so the +/- buttons and the edge-drag handles on the scrollbar all clamp to bounds that move with the project. Re-clamping fires whenever those bounds shift — adding a long clip or resizing the window pulls the current zoom inside the new range instead of leaving the user parked outside it.

* fix(stories): show the source filename on imported clips

Imports were rendering as "Imported Audio" everywhere because every
import points at the singleton voice profile. The filename was already
being stored on the generation row (in the `text` field), so the chat
item title and the timeline clip label now read from `text` when
`engine === 'import'` and fall back to the profile name otherwise. The
chat item also drops the language pill (always "en" on imports — not
informative) and skips the transcript textarea since imports have no
spoken text to show.

* fix(stories): round split_time_ms before posting

handleSplit was sending currentTimeMs - item.start_time_ms straight to the backend, which rejects it because StoryItemSplit.split_time_ms is typed as int and the playhead's currentTimeMs is a float (it's driven from HTMLAudioElement.currentTime, which carries sub-millisecond precision). Pydantic surfaced the mismatch as "Input should be a valid integer, got a number with a fractional part" and the toast read "Failed to split clip". Math.round at the call site, matching what the trim and move handlers already do.

* feat(stories): per-clip volume control on the timeline

Each story item now carries a volume column (linear gain, default 1.0,
clamped 0.0–2.0 server-side). New PUT /stories/{}/items/{}/volume route
+ useUpdateStoryItemVolume hook + a Volume2 icon in the clip-edit
toolbar that opens a popover with a 0–200% slider. Local slider state
drives the visual during a drag; the persist fires once on
onValueCommit, mirroring the generation-page slider pattern.

Web Audio playback inserts a per-clip GainNode between source and
master so volume changes apply live without re-decoding the buffer
(source -> clipGain -> masterGain -> destination). Server-side
mixdown in export multiplies the trimmed clip by its volume before
summing into the timeline. Split + duplicate carry the volume forward
to the new clips so trimming a faded section keeps the level you set.

Migration adds the volume column with default 1.0 so existing rows
read as full volume.

* fix(stories): mute the clip waveform's media element so it can't bleed audio

The clip waveforms drawn inside each timeline track use WaveSurfer with the default MediaElement backend, which creates an internal <audio> element to drive playback timing. Web Audio in useStoryPlayback is what actually produces sound, but WaveSurfer's element was happily preloading and — after the first user gesture unlocked browser autoplay — playing the source URL through the page output too.

For TTS clips it was masked: they're short, both sources start at the same time, and stopping the BufferSourceNode at pause coincides with the natural end of the audio element. For long imports (a four-minute MP3) the BufferSourceNode stops on pause but WaveSurfer's element keeps going on its own track — which is exactly the "music keeps playing when I pause" symptom.

Hand WaveSurfer a muted <audio> element via the `media` option so the visual still loads peaks but the element itself can never produce sound. preload="metadata" keeps the load lightweight.

* fix(stories): hard-cut the audio graph on stop so long imports actually halt

source.stop() was the only thing happening when a clip was halted, and on long imported buffers (multi-minute MP3s scheduled via source.start with a duration argument) it was silently failing to halt the buffer in some browsers — pause left the music playing and seek stacked another source on top of the original. The mute-the-WaveSurfer-element fix was a different bug along the same path; this is the one that actually addresses the duplicated audio.

ActiveSource now carries the per-clip GainNode alongside the source, and stopSource detaches the onended handler before calling stop() (so the natural-end callback can't race with explicit teardown and re-delete a freshly rescheduled entry at the same id), then disconnects both nodes inside their own try/catch blocks. Even when stop() doesn't actually halt the buffer the graph is severed — no path from source to destination, no audio.

* feat(stories): add empty tracks above/below the timeline

Tiny + strips sit at the top of the topmost label cell and the bottom of the bottommost one, sticky-positioned in the label column so they follow horizontal scroll. Clicking either extends the visible track stack in that direction by one — above adds max(existing)+1, below adds min(existing)-1. Both compute against the full set (defaults + item-derived + previously-added) so successive clicks keep extending instead of fighting over the same number.

Empty extras live in component state because a track only earns its keep once a clip lands on it. Once one does, item.track carries the number forward and the row keeps deriving from items naturally; if nothing lands there before reload, the empty row simply isn't there next time, which matches what the user expects of an unused affordance.

* fix(mcp): bundle stdio shim sidecar

* fix(captures): allow dictation without paste permission

* fix(mcp): preserve speak engine defaults

* fix(captures): use platform hotkey defaults

* fix(mcp): preload speak pill window

* fix(captures): hide unwired storage settings

* feat(sponsors): add /sponsors page, homepage promo, and in-app strip

* style(landing): drop pill chrome from /download maintainer kicker

* changelog

* better naming for sponsors

* windows keybind note

---------

Co-authored-by: Claude Opus 4.7 (1M context) <[email protected]>
This commit is contained in:
Jamie Pine
2026-04-25 15:46:35 -07:00
committed by GitHub
co-authored by Claude Opus 4.7
parent 627d40b42d
commit 7df366d0c8
171 changed files with 20611 additions and 1293 deletions
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---
title: "Voicebox Documentation"
description: "Voicebox is a local-first voice cloning studio -- a free and open-source alternative to ElevenLabs."
description: "Voicebox is the open-source, local-first AI voice studio a free alternative to ElevenLabs and WisprFlow, running entirely on your machine."
---
Voicebox is a **local-first voice cloning studio** -- a free and open-source alternative to ElevenLabs. Clone voices from a few seconds of audio, generate speech in 23 languages across 7 TTS engines, apply post-processing effects, and compose multi-voice projects with a timeline editor.
Voicebox is the **open-source, local-first AI voice studio** — a free
alternative to ElevenLabs and WisprFlow in one app. Clone voices, generate
speech across 7 TTS engines, dictate into any app with a global hotkey,
compose multi-voice projects, and let any MCP-aware agent speak in a voice
you own. Everything runs on your hardware.
![Voicebox App Screenshot](/images/app-screenshot-1.webp)
- **Complete privacy** -- models and voice data stay on your machine
- **7 TTS engines** -- Qwen3-TTS, Qwen CustomVoice, LuxTTS, Chatterbox Multilingual, Chatterbox Turbo, HumeAI TADA, and Kokoro
- **Cloning and preset voices** -- zero-shot cloning from a reference sample, or 50+ curated preset voices via Kokoro and Qwen CustomVoice
- **23 languages** -- from English to Arabic, Japanese, Hindi, Swahili, and more
- **Post-processing effects** -- pitch shift, reverb, delay, chorus, compression, and filters
- **Expressive speech** -- paralinguistic tags like `[laugh]`, `[sigh]`, `[gasp]` via Chatterbox Turbo; natural-language delivery control via Qwen CustomVoice
- **Unlimited length** -- auto-chunking with crossfade for scripts, articles, and chapters
- **Stories editor** -- multi-track timeline for conversations, podcasts, and narratives
- **API-first** -- REST API for integrating voice synthesis into your own projects
- **Native performance** -- built with Tauri (Rust), not Electron
- **Runs everywhere** -- macOS (MLX/Metal), Windows (CUDA), Linux, AMD ROCm, Intel Arc, Docker
- **Dictation** — hold a chord anywhere on your machine, speak, release; the transcript pastes into the focused field
- **Captures tab** — paired audio + transcript archive, retranscribe / refine / play-as-voice
- **Voice personalities** — per-profile compose button + persona-rewrite toggle, powered by a local LLM
- **Agents speak back** — any MCP-aware agent can call Voicebox to speak in one of your cloned voices
- **7 TTS engines** — Qwen3-TTS, Qwen CustomVoice, LuxTTS, Chatterbox Multilingual, Chatterbox Turbo, HumeAI TADA, Kokoro
- **Cloning and preset voices** — zero-shot cloning or 50+ curated preset voices
- **23 languages** — from English to Arabic, Japanese, Hindi, Swahili
- **Post-processing effects** — pitch shift, reverb, delay, chorus, compression, filters
- **Expressive speech** — paralinguistic tags (`[laugh]`, `[sigh]`) and natural-language delivery control
- **Unlimited length** — auto-chunking with crossfade for long scripts
- **Stories editor** — multi-track timeline for conversations, podcasts, narratives
- **API-first** — REST + WebSocket API, MCP server for agent integrations
- **Complete privacy** — models, audio, transcripts, LLM output never leave your machine
- **Runs everywhere** — macOS (MLX/Metal), Windows (CUDA / DirectML), Linux (ROCm / CPU), Intel Arc, Docker
## Download
@@ -32,6 +39,8 @@ Voicebox is a **local-first voice cloning studio** -- a free and open-source alt
## Get Started
- [Installation](/overview/installation) -- download and install Voicebox
- [Quick Start](/overview/quick-start) -- get up and running in 5 minutes
- [API Reference](/api-reference) -- integrate voice synthesis into your apps
- [Installation](/overview/installation) download and install Voicebox
- [Quick Start](/overview/quick-start) get up and running in 5 minutes
- [Dictation](/overview/dictation) — start talking to your computer
- [Voice Personalities](/overview/voice-personalities) — compose and rewrite in any profile
- [API Reference](/api-reference) — integrate voice synthesis into your apps
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---
title: "Captures"
description: "The paired audio + transcript archive — every dictation, recording, and uploaded audio file shows up here, replayable and retranscribable."
---
## Overview
A **capture** is an audio clip paired with its transcript. The Captures tab
is where every dictation, manual recording, and uploaded audio file lands,
with the original audio kept alongside the text so you can replay, re-run
transcription with a different model, refine the transcript, or send the
content somewhere else — including generating it back as speech in any of
your voice profiles.
<Callout type="info">
The Captures tab shipped in **0.5.0**, alongside global dictation and the
per-profile personality modes. If you've used earlier versions, note that
the Audio tab moved into **Settings → Audio Channels** to make room for
this one.
</Callout>
## Where captures come from
| Source | How it shows up | Badge |
|---|---|---|
| **Dictation** | Triggered by the global hotkey (see [Dictation](/overview/dictation)). Auto-refined by default. | `dictation` |
| **In-app recording** | Recorded directly in the Captures tab using the built-in mic. | `recording` |
| **File upload** | Any audio file dropped into the Captures tab — `.wav`, `.mp3`, `.m4a`, `.webm`, `.opus`, `.flac`. | `file` |
All three paths share the same backend pipeline, the same model picker, and
the same refinement flags. The source badge is there so you can visually
scan a long list.
## List view
The main Captures view is a chronological list. Each row shows:
- The transcript (raw or refined — the refined version wins if present)
- Duration + timestamp
- Source badge
- A play button for the original audio
- A meatballs menu with per-row actions
Filtering and search are a Tier-2 ask — ping if you need them.
## Detail view
Clicking into a capture opens the detail view:
- **Waveform player** for the original audio
- **Transcript editor** — click in and edit. Changes save on blur.
- **Refined vs. raw toggle** if refinement ran on this capture
- **Per-capture action bar** — retranscribe, refine, play as voice, delete
- **Settings snapshot** — STT model used, refinement flags at the time
this capture was processed, and the voice model if any was played
## Retranscribe
Runs the capture's original audio through a different Whisper model without
re-uploading or re-refining anything. Useful when:
- The default model mis-heard something and you want to try a larger model
- You used Base for a noisy clip and want to rerun with Turbo
- A non-English clip needs an explicit language hint
**Settings → Captures → Transcription** controls the default model and
language lock for new captures. Retranscribe uses those defaults unless you
override them per capture.
## Refine
Runs the raw transcript through the local LLM to produce a cleaned-up
version. The flags on the capture are snapshotted when refinement first
runs, so you can re-refine later with different flags without losing the raw
transcript:
| Flag | Effect |
|---|---|
| **Smart cleanup** | Remove fillers (`um`, `uh`, `like`), tidy punctuation and capitalization. |
| **Remove self-corrections** | Keep the final version when the speaker backtracks ("actually, no, on Tuesday"). |
| **Preserve technical terms** | Leave identifiers (`handleSubmit`, `npm install`) untouched. |
See the Refinement section of [Dictation](/overview/dictation#refinement) for
how Voicebox strips Whisper loop hallucinations *before* the LLM sees the
transcript — a capture can be re-refined any number of times without
re-introducing "thanks for watching thanks for watching" echoes.
The refinement model picker (three bundled Qwen3 sizes) lives in
**Settings → Captures → Refinement**.
## Play as voice
This is the capability no one else in the dictation category ships: take any
capture and play it back as speech in any of your voice profiles. One
dropdown over every profile, one click, and the capture's text runs through
`/generate` with the selected voice.
Use cases:
- Hear your own dictation back in a cloned voice of someone you like
- Send a message you dictated as an audio reply in a specific character
- Quickly prototype a line for a story without retyping
Playback uses whatever engine the selected profile is bound to — the same
rules as the Generate tab. There's no LLM in this path; the transcript goes
through unchanged. If you want the agent-style "transform the content before
speaking" flow, that's what the
[personality modes](/overview/voice-personalities) do — and the same
primitive is exposed to MCP-aware agents via the
[MCP Server](/overview/mcp-server) so Claude Code, Cursor, or Cline can speak
in one of your voices on their own.
<Callout type="info">
The default voice for the Captures tab's Play-as action is set in
**Settings → Captures → Playback → Default voice**. You can still override
it per capture.
</Callout>
## Send-to menu
Each capture has a Send-to menu for moving its content into other parts of
Voicebox:
- **Copy transcript** — to clipboard
- **Use as voice sample…** — promote this capture to a sample on a voice
profile of your choice. Opens a profile picker (with "+ New voice" for
cold starts) and a reference-text confirm dialog, because cloning needs
the `reference_text` to match the audio verbatim. Edit as needed and
save — the capture stays in the Captures tab untouched; the sample is a
copy, not a move.
## Storage
The original audio is kept alongside the transcript in your Voicebox data
directory. **Settings → Captures → Storage** shows the captures folder and can
open it directly in your file manager.
Every capture's audio file and metadata row can be re-processed (retranscribe,
refine, Play-as) as long as the audio file still exists.
## Short-recording guard
Audio clips under **300 ms** are short-circuited client-side and never
uploaded. This prevents a fumbled chord tap from landing an empty capture.
The threshold is tuned to filter accidents without cutting off intentional
short dictations.
## Keyboard shortcuts
Inside the Captures tab:
| Keys | Action |
|---|---|
| `Space` | Play / pause the selected capture |
| `↑` / `↓` | Previous / next capture in the list |
| `Enter` | Open the selected capture in detail view |
| `⌘ / Ctrl` + `C` (in detail view) | Copy the transcript |
## API surface
The Captures tab is backed by a small set of REST endpoints:
| Method | Endpoint | Use |
|---|---|---|
| `POST` | `/captures` | Upload audio + start the pipeline (STT, optional refinement, archival). |
| `GET` | `/captures` | List captures. |
| `GET` | `/captures/{id}` | Fetch one capture. |
| `POST` | `/captures/{id}/retranscribe` | Rerun STT with a chosen model. |
| `POST` | `/captures/{id}/refine` | Rerun refinement with chosen flags. |
| `POST` | `/profiles/{id}/samples/from-capture/{capture_id}` | Promote a capture to a voice profile sample. |
These endpoints are stable and usable from your own scripts — see
[Remote Mode](/overview/remote-mode) for running Voicebox as a server the rest
of your machine can talk to.
## Next steps
<Cards>
<Card title="Dictation" href="/overview/dictation">
The global hotkey flow that feeds most captures.
</Card>
<Card title="Voice Personalities" href="/overview/voice-personalities">
Per-profile compose button and persona rewrite toggle for captures you
want to transform, not just transcribe.
</Card>
<Card title="Creating Voice Profiles" href="/overview/creating-voice-profiles">
Promote a capture into a voice sample on a profile.
</Card>
</Cards>
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---
title: "Dictation"
description: "Hold a key anywhere on your machine, speak, release — the transcript lands in whatever text field you had focused."
---
## Overview
Dictation lets you turn speech into clean text anywhere on your computer. Hold
a chord, talk, release — Voicebox transcribes what you said with Whisper,
optionally cleans it up with a local LLM, and pastes the result into the text
field you had focused when you started.
Everything happens on your hardware. No cloud, no accounts, no audio leaving
the machine.
<Callout type="info">
Dictation was introduced in **0.5.0** alongside the Captures tab and the
per-profile personality modes. It's the "input" half of Voicebox's voice I/O
loop — cloning and TTS are still the "output" half.
</Callout>
## The flow
<Steps>
<Step title="Hold the chord">
Hold the push-to-talk chord anywhere on your machine. A small pill fades
in over your current app.
</Step>
<Step title="Speak">
The pill shows `Recording` with a live waveform and an elapsed-time
counter. Speak naturally — you don't have to wait for anything.
</Step>
<Step title="Release">
On release, the pill flips to `Transcribing`, then `Refining` if
auto-refine is on, then disappears.
</Step>
<Step title="Text lands in your app">
If auto-paste is enabled and Voicebox has Accessibility permission, the
transcript pastes into the text field you had focused when you started
talking — not wherever focus drifted while you were speaking.
</Step>
</Steps>
Either way, every capture also appears in the **Captures tab** with the
original audio and the transcript paired together. See
[Captures](/overview/captures) for what you can do with them after the fact.
## Push-to-talk and toggle modes
Voicebox ships two chord behaviors out of the box:
| Mode | Default (macOS) | Default (Windows) | Behavior |
|---|---|---|---|
| **Push-to-talk** | Right `⌘` + Right `⌥` | Right `Ctrl` + Right `Shift` | Recording stops when you release the chord. |
| **Toggle-to-talk** | Push-to-talk + `Space` | Push-to-talk + `Space` | Recording keeps going until you tap the chord again. |
**Holding PTT and tapping `Space` mid-hold upgrades a hold into a toggled
session** without a gap in the audio. This is the single most useful detail of
the chord system — short bursts feel fast, long-form narration feels
hands-free, and there's no decision up front about which mode you wanted.
## The on-screen pill
While you're dictating, a floating pill appears over the current app. It walks
through the states of the capture cycle and shows live signals for each:
| State | What it shows |
|---|---|
| `Recording` | Live waveform + elapsed time. |
| `Transcribing` | Thinking waveform while Whisper runs. |
| `Refining` | Same thinking waveform while the LLM cleans up the transcript (only if auto-refine is on). |
| Error | Red tint. Click the pill to copy the error to your clipboard. Auto-dismisses. |
The pill is transparent, always-on-top, and pre-created hidden at app start —
so it appears instantly when you hit the chord, with no window flash.
## Customizing the chord
Open **Settings → Captures → Dictation** to change either chord.
- **Left vs right modifier badges.** When you hold keys into the chord
picker, Voicebox records whether each modifier is the left or right variant.
That means you can bind to just the right `⌥` while leaving the left `⌥`
alone — useful if you want dictation on one hand and keep your
other-hand shortcuts intact.
- **Chord defaults are picked to stay out of your way.** On macOS, the
defaults deliberately avoid left-hand `Cmd+Option` chords so
`Cmd+Option+I` (devtools), `Cmd+Option+Esc` (force quit), and
`Cmd+Option+Space` (Spotlight) all remain yours. On Windows, the defaults
route around AltGr collisions on German / French / Spanish layouts where
`Ctrl+Alt` synthesizes AltGr.
- **Live reload.** Changing a chord in Settings takes effect immediately —
no restart, no tab reload.
## Auto-paste into the focused app
Once transcription finishes, Voicebox can synthesize a native paste into
whatever text field had focus when you started the chord. Your clipboard is
saved before and restored after, so nothing you had copied goes missing.
| Platform | Mechanism |
|---|---|
| macOS | `CGEventPost` at the HID tap with a full `⌘V` key sequence, preceded by reactivating the original app via `NSRunningApplication`. |
| Windows | `SendInput` with correct scan codes, plus a `SetForegroundWindow` + `AttachThreadInput` handshake to defeat foreground-lock when pasting into a window that wasn't frontmost at chord-start. |
**Focus is snapshotted at chord-start.** The paste targets the original field
even if focus drifts during transcribe / refine — that's the "pastes where you
were talking *from*, not where you're looking *now*" behavior.
<Callout type="info">
Auto-paste is optional. If Accessibility permission isn't granted (macOS),
or you prefer to keep synthetic input off, dictation still runs — transcripts
land in the Captures tab and you can copy them manually. The setting lives
inline next to the Accessibility prompt in Settings → Captures → Dictation,
not as a global banner.
</Callout>
## Refinement
If auto-refine is on, a local LLM cleans up the raw Whisper transcript
before it's pasted. The goal is to remove verbal clutter without rewriting
what you actually said.
What refinement typically fixes:
- Filler words (`um`, `uh`, `like` used as pauses, `you know`)
- Self-corrections — the LLM keeps the final version and drops earlier
attempts (`could you uh run the migration real quick, and then, yeah,
check the logs` → `Could you run the migration, then check the logs?`)
- Basic punctuation and capitalization
- Whisper loop hallucinations — Voicebox strips repeated tokens (six or
more identical tokens in a row, case-insensitive) *before* the LLM
sees the transcript, so a small refinement model can't echo them back
What refinement deliberately preserves:
- Technical terms and code identifiers (`npm install`, `handleSubmit`)
- Legitimate repetition (`no, no, no, no, no` has fewer than six identical
tokens, so it survives)
- Your intent — refinement is cleanup, not rewriting
Flags are snapshotted per capture, so you can re-refine the same raw
transcript later with different flags without losing the original. The
refinement model picker (**Settings → Captures → Refinement**) offers three
bundled Qwen3 sizes:
| Model | Size | Best for |
|---|---|---|
| Qwen3 0.6B | ~400 MB | Default. Very fast, good for casual dictation. |
| Qwen3 1.7B | ~1.1 GB | Sweet spot when transcripts contain code identifiers. |
| Qwen3 4B | ~2.5 GB | Full quality, slowest. |
This is the same local LLM used by the per-profile personality modes — one
LLM in the app, not two. See [Voice Personalities](/overview/voice-personalities).
## Platform notes
### macOS
- **Accessibility permission** is required for auto-paste. The prompt lives
inline next to the toggle in **Settings → Captures → Dictation**, with a
deep link to **System Settings → Privacy & Security → Accessibility**.
- **TSM crash mitigation.** The global hotkey listener runs on a background
thread with `set_is_main_thread(false)` to sidestep a known
macOS 14+ crash in the `rdev` library. If you hit an unexpected dictation
failure on macOS, check the logs for TSM-related messages.
### Windows
- **UAC / UIPI caveat.** Synthetic paste into an *elevated* window from a
non-elevated Voicebox is blocked by Windows itself. Run Voicebox elevated
if you regularly dictate into elevated apps (e.g. an elevated terminal or
Task Manager).
- **Right-hand default chord** (`Ctrl+Shift`) avoids AltGr collisions on
keyboard layouts where `Ctrl+Alt` is the compose key (German, French,
Spanish, some others).
### Linux
- **Not yet in this release.** The Rust shim ships the macOS and Windows
paths in 0.5.0. Linux `uinput` / AT-SPI support and the Wayland paste
story are tracked in `docs/plans/VOICE_IO.md`.
## When auto-paste skips itself
A few cases where Voicebox deliberately does *not* synthesize a paste:
- **Focus was inside Voicebox** when the chord started. The transcript goes
to the Captures tab so a dictation-into-Voicebox round-trip doesn't
accidentally paste into the generate box.
- **No text focus detected.** The transcript still lands in the Captures
tab; copy it from there with one click.
- **Accessibility permission not granted** on macOS. Same — Captures tab
only.
## Next steps
<Cards>
<Card title="Captures" href="/overview/captures">
The paired audio + transcript archive every dictation lands in.
</Card>
<Card title="Voice Personalities" href="/overview/voice-personalities">
The same local LLM powers per-profile compose and persona rewrite.
</Card>
<Card title="Transcription" href="/developer/transcription">
Developer-level details on Whisper, Whisper Turbo, and the STT backend.
</Card>
</Cards>
+60 -20
View File
@@ -1,23 +1,48 @@
---
title: "Introduction"
description: "Voicebox is a local-first voice cloning studio -- a free and open-source alternative to ElevenLabs."
description: "Voicebox is the open-source, local-first AI voice studio a free alternative to ElevenLabs and WisprFlow, running entirely on your machine."
---
## What is Voicebox?
Voicebox is a **local-first voice cloning studio** -- a free and open-source alternative to ElevenLabs. Clone voices from a few seconds of audio or pick from 50+ preset voices, generate speech in 23 languages across 7 TTS engines, apply post-processing effects, and compose multi-voice projects with a timeline editor.
Voicebox is the **open-source, local-first AI voice studio**. It closes the
voice I/O loop in both directions on one machine, with no cloud and no
accounts:
- **Complete privacy** -- models and voice data stay on your machine
- **7 TTS engines** -- Qwen3-TTS, Qwen CustomVoice, LuxTTS, Chatterbox Multilingual, Chatterbox Turbo, HumeAI TADA, and Kokoro
- **Cloning and preset voices** -- zero-shot cloning from a reference sample, or curated preset voices via Kokoro (50 voices) and Qwen CustomVoice (9 voices)
- **23 languages** -- from English to Arabic, Japanese, Hindi, Swahili, and more
- **Post-processing effects** -- pitch shift, reverb, delay, chorus, compression, and filters
- **Expressive speech** -- paralinguistic tags like `[laugh]`, `[sigh]`, `[gasp]` via Chatterbox Turbo; natural-language delivery control via Qwen CustomVoice
- **Unlimited length** -- auto-chunking with crossfade for scripts, articles, and chapters
- **Stories editor** -- multi-track timeline for conversations, podcasts, and narratives
- **API-first** -- REST API for integrating voice synthesis into your own projects
- **Native performance** -- built with Tauri (Rust), not Electron
- **Runs everywhere** -- macOS (MLX/Metal), Windows (CUDA), Linux, AMD ROCm, Intel Arc, Docker
- **Humans talk** — hold a chord anywhere on your machine and your
dictation lands as clean text in whatever text field you had focused
- **Agents talk back** — any MCP-aware agent can call Voicebox to speak in
one of your cloned voices
- **Voices speak for themselves** — voice profiles can carry a personality
that composes fresh lines or rewrites text before it's spoken
It's the free, local alternative to both ElevenLabs (voice cloning and TTS)
and WisprFlow (voice dictation for agents and power users) — covering both
sides of the same loop in one app, with a single model directory and LLM
shared between input and output.
## What's in the app
- **Dictation** — global hotkey, push-to-talk and toggle modes, auto-paste
into the focused field on macOS and Windows (see [Dictation](/overview/dictation))
- **Captures tab** — paired audio + transcript archive, retranscribe,
refine, play-as-voice, promote-to-sample (see [Captures](/overview/captures))
- **Voice cloning** — 5 cloning engines covering 23 languages. Zero-shot
cloning from a reference sample (see [Voice Cloning](/overview/voice-cloning))
- **Preset voices** — 50+ curated voices via Kokoro and Qwen CustomVoice
for when you don't want to clone (see [Preset Voices](/overview/preset-voices))
- **Voice personalities** — optional free-form personality on any profile
plus a compose button and persona-rewrite toggle powered by a local LLM (see
[Voice Personalities](/overview/voice-personalities))
- **Post-processing effects** — pitch shift, reverb, delay, chorus,
compression, filters (Spotify's Pedalboard)
- **Expressive speech** — paralinguistic tags like `[laugh]` and `[sigh]`
via Chatterbox Turbo; natural-language delivery control via Qwen CustomVoice
- **Unlimited length** — auto-chunking with crossfade for long scripts
- **Stories editor** — multi-track timeline for conversations and podcasts
- **API-first** — REST + WebSocket API; MCP server for agent integrations
- **Runs everywhere** — macOS (MLX/Metal), Windows (CUDA / DirectML), Linux
(ROCm / CPU), Intel Arc, Docker
## TTS Engines
@@ -30,9 +55,21 @@ Seven engines with different strengths, switchable per-generation:
| **LuxTTS** | Cloned | English | Lightweight (~1GB VRAM), 48kHz output, 150x realtime on CPU |
| **Chatterbox Multilingual** | Cloned | 23 | Broadest language coverage |
| **Chatterbox Turbo** | Cloned | English | Fast 350M model with paralinguistic emotion/sound tags |
| **TADA** (1B / 3B) | Cloned | 10 | HumeAI speech-language model -- 700s+ coherent audio |
| **TADA** (1B / 3B) | Cloned | 10 | HumeAI speech-language model 700s+ coherent audio |
| **Kokoro** | Preset (50 voices) | 9 | 82M parameters, CPU realtime, lowest VRAM of any engine |
## STT and local LLM
Voicebox also runs a full speech recognition and local LLM stack, shared
between dictation, the Captures tab, and per-profile personality modes:
| Layer | Models |
|---|---|
| **STT** | Whisper Base / Small / Medium / Large / Turbo (PyTorch or MLX) |
| **LLM** | Qwen3 0.6B / 1.7B / 4B (refinement + per-profile compose / persona-rewrite) |
No cloud fallback, no bring-your-own-API-key. Local is the product.
## GPU Support
| Platform | Backend | Notes |
@@ -46,11 +83,13 @@ Seven engines with different strengths, switchable per-generation:
## Use Cases
- **Game development** -- generate dynamic dialogue for characters
- **Content creation** -- produce podcasts and video voiceovers
- **Accessibility** -- build text-to-speech tools for users who need them
- **Voice assistants** -- create custom voice interfaces
- **Production pipelines** -- automate voiceover workflows via the REST API
- **Dictation for humans and agents** — speak instead of type, in any app
- **Agent voice output** — any MCP-aware agent can speak in a cloned voice
- **Game development** — generate dynamic dialogue for characters
- **Content creation** — podcasts, video voiceovers, audiobooks
- **Accessibility** — speech-to-text for any field, TTS with a voice you own
- **Voice assistants** — custom voice interfaces without a cloud bill
- **Production pipelines** — automate voice workflows via the REST API
## Tech Stack
@@ -61,8 +100,9 @@ Seven engines with different strengths, switchable per-generation:
| State | Zustand, React Query |
| Backend | FastAPI (Python) |
| TTS Engines | Qwen3-TTS, Qwen CustomVoice, LuxTTS, Chatterbox, Chatterbox Turbo, TADA, Kokoro |
| STT | Whisper / Whisper Turbo (PyTorch or MLX) |
| Local LLM | Qwen3 0.6B / 1.7B / 4B (MLX or PyTorch) |
| Effects | Pedalboard (Spotify) |
| Transcription | Whisper / Whisper Turbo (PyTorch or MLX) |
| Inference | MLX (Apple Silicon) / PyTorch (CUDA/ROCm/XPU/CPU) |
| Database | SQLite |
| Audio | WaveSurfer.js, librosa |
+299
View File
@@ -0,0 +1,299 @@
---
title: "MCP Server"
description: "Let Claude Code, Cursor, Cline, or any MCP-aware agent speak in one of your cloned voices — locally, with no cloud."
---
## Overview
Voicebox ships a built-in **Model Context Protocol** server so local AI
agents can call your Voicebox install directly: speak text in a voice
profile, transcribe audio, and list captures or profiles. The server runs
inside the same process as the rest of Voicebox and is mounted at `/mcp`
over Streamable HTTP.
Agent asks to speak → Voicebox plays audio on your speakers → an on-screen
pill surfaces the voice name for the whole duration so you always see what's
coming out of your machine.
<Callout type="info">
MCP shipped in **0.5.0** alongside [Dictation](/overview/dictation) and
[Voice Personalities](/overview/voice-personalities). The design goal is
"local voice layer for every agent on your machine" — the same app that
captures your voice can generate a response in any voice profile you've
cloned.
</Callout>
## Quick install
### Claude Code
```
claude mcp add voicebox \
--transport http \
--url http://127.0.0.1:17493/mcp \
--header "X-Voicebox-Client-Id: claude-code"
```
### Cursor / Windsurf / VS Code MCP / any HTTP MCP client
Drop this into the client's MCP config (usually `.mcp.json` or a Settings UI):
```json
{
"mcpServers": {
"voicebox": {
"url": "http://127.0.0.1:17493/mcp",
"headers": { "X-Voicebox-Client-Id": "cursor" }
}
}
}
```
Change `cursor` to whatever name you want the binding to show up as in
Voicebox → Settings → MCP. The value is just an identifier for the
per-client voice binding — not a secret, not a credential.
### Clients that only speak stdio
A stdio shim binary `voicebox-mcp` is bundled with the desktop app. Point
the client at that binary's absolute path:
<Tabs items={["macOS", "Windows", "Linux"]}>
<Tab value="macOS">
```json
{
"mcpServers": {
"voicebox": {
"command": "/Applications/Voicebox.app/Contents/MacOS/voicebox-mcp",
"env": { "VOICEBOX_CLIENT_ID": "claude-desktop" }
}
}
}
```
</Tab>
<Tab value="Windows">
```json
{
"mcpServers": {
"voicebox": {
"command": "C:\\Program Files\\Voicebox\\voicebox-mcp.exe",
"env": { "VOICEBOX_CLIENT_ID": "claude-desktop" }
}
}
}
```
</Tab>
<Tab value="Linux">
```json
{
"mcpServers": {
"voicebox": {
"command": "/opt/voicebox/voicebox-mcp",
"env": { "VOICEBOX_CLIENT_ID": "claude-desktop" }
}
}
}
```
</Tab>
</Tabs>
The shim waits up to 30 seconds for the Voicebox backend to come up, then
proxies JSON-RPC from stdio over Streamable HTTP. Voicebox must be running
for the shim to connect.
## Tools
| Tool | Use |
|---|---|
| `voicebox.speak` | Speak text in a voice profile. Returns a `generation_id` to poll. |
| `voicebox.transcribe` | Whisper transcription of base64 audio or an absolute local path. |
| `voicebox.list_captures` | Recent captures with transcripts, paginated. |
| `voicebox.list_profiles` | Available voice profiles (cloned + preset). |
### `voicebox.speak`
```ts
voicebox.speak({
text: "Deploy complete.",
profile?: "Morgan", // name or id; falls back to per-client binding, then default
engine?: "qwen", // qwen | qwen_custom_voice | luxtts | chatterbox | chatterbox_turbo | tada | kokoro
personality?: true, // rewrite via the profile's personality LLM before TTS; default comes from the per-client binding
language?: "en",
})
```
Returns:
```json
{
"generation_id": "…",
"status": "generating",
"profile": "Morgan",
"source": "mcp",
"poll_url": "/generate/<id>/status"
}
```
- **Plain TTS** — `personality: false` (or omitted + binding default is false). Text is spoken as-is.
- **Persona mode** — `personality: true` and the profile must have a personality prompt set.
The LLM rewrites the text in character before TTS. See [Voice Personalities](/overview/voice-personalities).
### `voicebox.transcribe`
```ts
voicebox.transcribe({
audio_base64?: "<base64>", // exactly one of these two
audio_path?: "/absolute/path/to/file.wav",
language?: "en",
model?: "turbo", // base | small | medium | large | turbo
})
```
Returns `{ text, duration, language, model }`. 200 MB ceiling on either path.
### `voicebox.list_captures`
`{ limit?: 20, offset?: 0 }` → `{ captures: [...], total }`. `limit` is
clamped to `1..=200`.
### `voicebox.list_profiles`
No args → `{ profiles: [{ id, name, voice_type, language, has_personality }] }`.
## Voice resolution
Every call to `voicebox.speak` (and `POST /speak`) resolves the voice profile
in this order:
<Steps>
<Step title="Explicit `profile` arg">
Passed as a name (case-insensitive) or id. If the name/id doesn't match,
the call errors — the server doesn't silently fall back.
</Step>
<Step title="Per-client binding">
Looked up by the `X-Voicebox-Client-Id` header. Managed in
**Voicebox → Settings → MCP**. Lets you pin Claude Code to Morgan,
Cursor to Scarlett, etc.
</Step>
<Step title="Global default">
`capture_settings.default_playback_voice_id` — same default voice the
Captures tab's "Play as voice" action uses.
</Step>
</Steps>
If none of the three produce a profile the tool returns a helpful error
pointing at Settings.
## Per-client bindings
Voicebox → Settings → MCP shows one row per `client_id` Voicebox has heard
from, plus the config snippets you can copy into each agent. Each row
carries:
| Field | Purpose |
|---|---|
| `label` | Display name in the Settings UI (e.g. "Claude Code"). |
| `profile_id` | The voice this client uses when `profile` isn't passed. |
| `default_engine` | Override the TTS engine for this client. |
| `default_personality` | When true, `voicebox.speak` routes through the profile's personality LLM (rewrite) by default. |
| `last_seen_at` | Last time the server saw a request from this client. |
`last_seen_at` is stamped automatically by middleware on every `/mcp/*`
request — useful when you're not sure whether your config took.
## The speaking pill
Every agent-initiated speak surfaces the floating pill the same way
[Dictation](/overview/dictation) does, in a new `Speaking` state showing the
profile name and an elapsed timer. The pill is intentionally unmissable —
silent background TTS is a trust hazard, so Voicebox always shows what's
being spoken and in what voice.
Behind the scenes, the backend broadcasts `speak-start` and `speak-end`
events on `GET /events/speak`, which `DictateWindow` subscribes to via SSE.
The pill overrides the capture session when both would render — you can't
hear two pills at once.
## Non-MCP REST surface
`POST /speak` is a thin wrapper on the same code path for callers that
don't speak MCP — shell scripts, ACP, A2A, GitHub Actions, whatever.
```bash
curl -X POST http://127.0.0.1:17493/speak \
-H 'Content-Type: application/json' \
-H 'X-Voicebox-Client-Id: ci' \
-d '{"text":"Build complete.","profile":"Morgan"}'
```
Body fields match the MCP tool: `text`, optional `profile`, `engine`,
`personality`, `language`. Returns a `GenerationResponse` — the same shape as
`POST /generate`.
## Debugging
Use the MCP Inspector to poke tools directly without plumbing through an
agent:
```
npx @modelcontextprotocol/inspector http://127.0.0.1:17493/mcp
```
Start with `voicebox.list_profiles` to confirm wiring, then
`voicebox.speak` for end-to-end — you should hear audio and see the
generation land in the Captures tab.
<Callout type="info">
If an agent can't reach the server, the first thing to check is that
Voicebox is running — the backend only listens while the desktop app is
open. The stdio shim surfaces this as a JSON-RPC error on the client
side after its 30-second health-wait window elapses.
</Callout>
## Security
- **Localhost only.** The server binds to `127.0.0.1`. If you ever point
Voicebox at a non-loopback interface (e.g. remote-mode over a trusted
network), add a bearer token — it's on the roadmap but not in 0.5.0.
- **No auth today.** Any process that can connect to your loopback can
call MCP. That's the same trust boundary as the rest of Voicebox's REST
API and is appropriate for a single-user local tool.
- **`audio_path` reads are unrestricted** against the same trust
boundary. If you're scripting against a shared host, prefer
`audio_base64` so you don't have to think about path sandboxing.
- **Voice cloning consent applies.** See [Voice Cloning](/overview/voice-cloning#limitations)
— an agent being able to call `voicebox.speak` in someone's voice
doesn't change the ethics of whose voices you clone.
## Implementation notes
- **Transport:** Streamable HTTP (Nov-2025 MCP spec, post-SSE). Claude
Code, Cursor, Windsurf, and VS Code MCP extensions all support it.
- **Package naming:** the backend package is `backend/mcp_server/`, not
`mcp`, to avoid shadowing the PyPI `mcp` package FastMCP imports
internally.
- **Dependencies:** `fastmcp>=3.0,<4.0`, `sse-starlette>=2.0`.
- **Lifespan:** mounting FastMCP requires the `lifespan=` kwarg on
`FastAPI()` — the startup/shutdown event decorators are incompatible
with FastMCP's Streamable HTTP session manager. The Voicebox app.py
composes both into one async context manager.
For the full developer-facing tour of the code layout, see
`backend/mcp_server/README.md` in the repo.
## Next steps
<Cards>
<Card title="Voice Personalities" href="/overview/voice-personalities">
Persona mode (`personality: true`) for agents that should
transform text in-character before speaking.
</Card>
<Card title="Dictation" href="/overview/dictation">
The pill that surfaces agent speech is the same one that surfaces
your dictations — one mental model for both directions of the loop.
</Card>
<Card title="Captures" href="/overview/captures">
Every agent-initiated speak lands in the Captures tab with its
generated audio — replay, download, repurpose.
</Card>
</Cards>
+4
View File
@@ -7,8 +7,12 @@
"docker",
"quick-start",
"gpu-acceleration",
"dictation",
"captures",
"voice-cloning",
"preset-voices",
"voice-personalities",
"mcp-server",
"stories-editor",
"recording-transcription",
"generation-history",
@@ -1,64 +1,106 @@
---
title: "Recording & Transcription"
description: "Record audio and transcribe speech with Whisper"
description: "A map of the three places you can record and transcribe audio in Voicebox — dictation, captures, and voice-profile samples."
---
## Recording
## Overview
Voicebox includes built-in recording capabilities for creating voice samples and capturing audio.
Voicebox records and transcribes audio in three different contexts, each
feeding a different surface in the app. This page is a map; follow the links
for the detail.
### Features
| Goal | Where | Docs |
|---|---|---|
| Speak and have your words land in another app | Global hotkey → Captures tab + auto-paste | [Dictation](/overview/dictation) |
| Record a thought, a meeting, or a voice memo inside Voicebox | Captures tab | [Captures](/overview/captures) |
| Record a clip to clone a voice from | Voices tab → profile samples | [Creating Voice Profiles](/overview/creating-voice-profiles) |
- **Microphone input** - Record from any audio input device
- **System audio capture** - Record desktop audio (macOS/Windows)
- **Waveform visualization** - See audio levels in real-time
- **Multiple formats** - Export as WAV, MP3, or M4A
All three paths share the same STT backend — it's the surrounding workflow
that differs.
### How to Record
## Dictation
<Steps>
<Step title="Select Input">
Choose your microphone or system audio
</Step>
<Step title="Start Recording">
Click the record button and speak clearly
</Step>
<Step title="Stop & Save">
Click stop when finished
</Step>
<Step title="Use or Export">
Use as voice sample or export to file
</Step>
</Steps>
The 0.5.0 headline feature. Hold a chord anywhere on your machine, speak,
release. The transcript lands in whatever text field you had focused,
cleaned up by a local LLM if auto-refine is on. Captures accumulate in the
Captures tab for later replay or re-transcription.
## Transcription
Covered end-to-end in [Dictation](/overview/dictation).
Automatic speech-to-text powered by OpenAI's Whisper model.
## Captures tab
### Features
When you don't need to paste into another app — you just want a clean
transcript of some audio — the Captures tab is the home. Record in-app,
drop in a file (`.wav`, `.mp3`, `.m4a`, `.webm`, `.opus`, `.flac`), or dig
through dictations that already landed there. Every capture keeps its
original audio, can be retranscribed with a different model, and can be
played back through any voice profile you have.
- **High accuracy** - Industry-leading speech recognition
- **Multiple languages** - Supports 50+ languages
- **Automatic detection** - Language auto-detection
- **Timestamps** - Word-level timing information
Covered in [Captures](/overview/captures).
### How to Transcribe
## Voice profile samples
<Steps>
<Step title="Select Audio">
Choose a recording or upload an audio file
</Step>
<Step title="Choose Language">
Select language or use auto-detect
</Step>
<Step title="Transcribe">
Click transcribe and wait for processing
</Step>
<Step title="Review & Export">
Review text and export as needed
</Step>
</Steps>
A separate flow, in the Voices tab. When you're creating a profile from an
audio clip, the sample is what the cloning engine actually learns from —
the `reference_text` on a sample must match the audio *verbatim*, which is
why samples are a different data model from captures.
You can promote a capture to a sample from the Captures tab's Send-to menu
("Use as voice sample…"), which opens a reference-text confirm dialog so
you can correct the last ~10% of transcript accuracy before saving.
Covered in [Creating Voice Profiles](/overview/creating-voice-profiles).
## Transcription models
All three paths share the same Whisper models. Pick a default in
**Settings → Captures → Transcription**; override per capture if you need
to.
| Model | Size | When to pick it |
|---|---|---|
| Whisper Base | ~300 MB | Fast. Default. Good for clean speech. |
| Whisper Small | ~500 MB | Better quality, still fast. |
| Whisper Medium | ~1.5 GB | High quality. |
| Whisper Large | ~3 GB | Best quality, slow on CPU. |
| Whisper Turbo | ~1.5 GB | Large-tier quality, ~5× faster than Large. |
On Apple Silicon the model runs through **MLX-Whisper** (~8× faster than
PyTorch). Everywhere else it runs through PyTorch `transformers`. The
backend picks the right one — you don't configure it.
<Callout type="info">
Transcription is useful for creating voice samples from existing audio or generating subtitles.
For noisy clips, prefer **Turbo** or **Large**. Base can hallucinate on
hard inputs — most famously the "thanks for watching" loop. Voicebox
strips those loops deterministically before LLM refinement runs, so a
capture can be cleanly re-refined even if the raw transcript has them.
</Callout>
## Language
You can pass a language hint for short clips (under ~5 seconds) where
Whisper's auto-detect is unreliable. Set a default language lock in
**Settings → Captures → Transcription → Language**, or override per capture.
## Transcription API
Developer-level detail on the STT backend, model loading, preprocessing, and
the `/transcribe` endpoint lives in the
[Transcription developer guide](/developer/transcription). The Captures
pipeline also exposes `/captures` as a higher-level endpoint that wraps
STT + archival + optional refinement in one call — see
[Captures](/overview/captures#api-surface).
## Next steps
<Cards>
<Card title="Dictation" href="/overview/dictation">
Hold a chord anywhere on your machine, speak, release.
</Card>
<Card title="Captures" href="/overview/captures">
The paired audio + transcript archive.
</Card>
<Card title="Creating Voice Profiles" href="/overview/creating-voice-profiles">
Record or upload samples for voice cloning.
</Card>
</Cards>
@@ -0,0 +1,186 @@
---
title: "Voice Personalities"
description: "Attach a personality to a voice profile, compose fresh in-character lines, and rewrite input text in their voice — all powered by a local LLM."
---
## Overview
A **personality** is an optional free-form description attached to a voice
profile — who this voice is, how they speak, what they care about. Set one
and two new controls appear next to the generate button, both powered by a
bundled Qwen3 LLM running entirely locally:
- **Compose** — drop a fresh in-character line into the textarea. Click
again for a different take.
- **Speak in character** — a toggle that rewrites your input text in the
character's voice before TTS, preserving every idea.
The LLM produces the text. The voice profile speaks it. No cloud round-trip,
no external API — the whole loop runs on your hardware.
<Callout type="info">
Personalities shipped in **0.5.0**. The same local LLM doubles as the
refinement model for [Dictation](/overview/dictation) — one LLM in the app,
not two, sharing one model cache and one GPU-memory footprint.
</Callout>
## Setting a personality
Open a voice profile's edit view. The **Personality** field is free-form text
up to **2,000 characters**. Describe the voice however helps you — past
lines they'd say, speech patterns, tone, boundaries.
Good descriptions tend to include:
- A one-line identity (who they are)
- Speech patterns (rhythm, vocabulary, what they avoid)
- Representative phrases — example lines show the LLM the target tone
better than adjectives
- What the character *wouldn't* do (they don't explain, they don't
apologize, they refuse to break character, etc.)
You can set a personality on any voice profile type — cloned or preset. The
three modes work identically regardless of engine.
## The two actions
Each action is tuned for a specific job and the LLM temperature is adjusted
to match.
### Compose
Generate a fresh utterance in the character's voice, with no seed text.
Click the shuffle button to drop a line straight into the generate
textarea; click again for a different take.
- **When to use:** prototyping, sampling a character's voice, brainstorming
a line without typing one first
- **Temperature:** hot — variety is the point
- **Typical output:** a short, punchy line that fits the character's
register
### Speak in character (rewrite)
Flip the persona toggle and whatever you type (or dictate) gets rewritten in
the character's voice before TTS — every idea preserved, only the phrasing
changes. High-fidelity mode: the content doesn't change, only the voice does.
- **When to use:** turning a dictated memo into in-character speech; lifting
a plain-English script into a specific voice without editing by hand
- **Temperature:** cold — faithfulness wins
- **Typical output:** same ideas, same order, different phrasing and cadence
## Speech-only framing
Both modes enforce **speech-only** output. The LLM is prompted to
produce things a person would actually say out loud — no narration, no
action tags (`*sighs*`, `[laughs]`), no meta-commentary, no markdown
formatting, no stage directions.
This is deliberate: the output is going straight into TTS, and anything that
isn't speakable ends up either ignored or read literally. The speech-only
framing also makes the output land cleanly inside dialogue, so you can drop
a Respond result straight into a Story.
## The local LLM
The bundled LLM is **Qwen3**, available in three sizes:
| Model | Download size | Best for |
|---|---|---|
| Qwen3 0.6B | ~400 MB | Default. Very fast, good for casual use. |
| Qwen3 1.7B | ~1.1 GB | Sweet spot for character personalities with specific phrasing. |
| Qwen3 4B | ~2.5 GB | Full quality. Slowest. Useful for very particular tone. |
The model runs through the same backend split Voicebox already uses for TTS
— **MLX** (4-bit community quants) on Apple Silicon, **PyTorch** (transformers
`AutoModelForCausalLM`) everywhere else. Downloads go through the same cache
and model-management UI as TTS models.
Pick a size in **Settings → Captures → Refinement → Refinement model** — the
personality modes reuse it. If you switch models, both refinement and
personality output pick up the change on the next call.
## Using the controls
Both controls appear on the floating generate box when the selected profile
has a personality set.
<Steps>
<Step title="Compose — shuffle a line">
Click the shuffle button. The LLM runs and the result fills the generate
textarea. Edit if you want, then hit generate.
</Step>
<Step title="Speak in character — toggle persona rewrite">
Type (or dictate) what you want said. Flip the wand toggle on. Hit
generate — Voicebox runs the text through the personality LLM first,
then TTS speaks the rewritten version. Leave the toggle off for plain
TTS.
</Step>
</Steps>
Compose always gives you something different on re-click. The persona
toggle, on the other hand, is a mode — it applies to every generate call
until you flip it back off.
## Use cases
- **Agents that speak in a voice you own.** Combine the persona toggle with
the built-in [MCP Server](/overview/mcp-server) so Claude Code, Cursor,
Cline, or any MCP-aware agent can talk back through a profile with a
personality. The agent calls `voicebox.speak({ text, profile, personality:
true })` and Voicebox rewrites the text in character before speaking.
- **Interactive characters.** Games, narrative tools, accessibility
experiences. A character with a personality description plus a cloned
voice becomes a reusable prop.
- **Accessibility.** People who can't speak in their original voice can
keep a personality description of how they used to sound and use the
rewrite toggle to turn typed input into in-character speech.
- **Creative drafting.** Write a plain outline, flip the persona toggle,
generate line-by-line into the character's voice, drop the audio into a
Story.
## API surface
Personalities are accessible via REST:
| Method | Endpoint | Body |
|---|---|---|
| `PUT` | `/profiles/{id}` | Include a `personality` field up to 2,000 chars to set it. |
| `POST` | `/profiles/{id}/compose` | No body. Returns a fresh in-character utterance as text. |
| `POST` | `/generate` | Include `personality: true` to run input text through the personality LLM before TTS. Same for `POST /speak`. |
`POST /generate` with `personality: true` is the same primitive MCP's
`voicebox.speak` tool uses when you pass `personality: true`. Scripts and
agents can use it directly.
## Limits and gotchas
- **The personality is a prompt, not a fine-tune.** The LLM will sometimes
drift out of character, especially on Compose at high temperature. Click
again for another take.
- **Long personalities are not always better.** 2,000 chars is a ceiling,
not a goal. A sharp 300-char description with two example lines
typically outperforms a long one.
- **Speech-only framing is enforced, but not bulletproof.** Very large
prompts or unusual inputs can sneak an action tag through. If you see
`[laughs]` in TTS output, it's usually a personality-field hint the
model anchored onto — remove it from the description.
- **Rewrite is stricter than Respond.** If the output is changing your
meaning, you probably want Respond (or a wholesale Compose with context
in the input), not Rewrite.
## Next steps
<Cards>
<Card title="Dictation" href="/overview/dictation">
Dictate the input for Rewrite or Respond from anywhere on your machine.
</Card>
<Card title="Captures" href="/overview/captures">
Captures feed personalities naturally — dictate a memo, rewrite it in
a character voice, generate speech.
</Card>
<Card title="Creating Voice Profiles" href="/overview/creating-voice-profiles">
Add a personality to an existing profile.
</Card>
</Cards>