The live mic-level halo around the stop button was frozen at a constant
scale (1.15) whenever the OS "Reduce motion" setting was on, so it never
reacted to the voice while dictating. Make haloScale unconditional so it
always follows audioLevel (amplitude 0.9), and drop the now-unused
useReducedMotion import and reduceMotion local.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Add a pulsing halo behind the stop button that scales with the
microphone input level, giving real-time feedback that recording is
active and the mic is picking up sound.
- use-dictation: meter the captured MediaStream via AudioContext +
AnalyserNode (analyser only, never connected to destination), compute
a smoothed RMS audioLevel (0..1) in a requestAnimationFrame loop, and
tear the meter down on every recording-end path (stop/cancel/auto-stop/
unmount); meter failure is non-fatal to recording
- mic-button: render a translucent red halo whose scale follows
audioLevel; honor prefers-reduced-motion with a static halo
- stop(): recover and release resources when no live recorder remains
- fix unhandled rejection from AudioContext.resume()
- ai.service: route *.openrouter.ai STT to its JSON+base64
/audio/transcriptions API; keep the OpenAI multipart path (AI SDK) for
OpenAI/self-hosted whisper. Unify transcription behind transcribe().
- /transcribe controller: surface the real provider/transport reason
(describeProviderError) instead of an opaque 500; preserve HttpException.
- testConnection: add an 'stt' capability (silent-WAV probe) + DTO; client
gets a Test endpoint button and status dot on the Voice/STT card.
- useDictation: log full errors to the console and show the real reason
(mic start + transcription paths); handle NotReadable/Abort and missing
mediaDevices.
- docs(CLAUDE.md): require full error logging + specific user-facing messages.
Add push-to-talk voice dictation that transcribes recorded audio on the
server via the workspace's OpenAI-compatible AI provider (Whisper /
gpt-4o-transcribe / self-hosted whisper), then inserts the text.
Backend:
- New `stt_api_key_enc` column + migration; STT creds parity with chat/
embeddings (sttModel/sttBaseUrl/sttApiKey, write-only key, fallbacks to
chat baseUrl/key). Both provider whitelists updated (service + repo).
- AiService.getTranscriptionModel + AiTranscriptionService.
- Gated POST /ai-chat/transcribe (dictation flag → 403, JWT + workspace
scope + throttle, 25MB cap, MIME whitelist, never logs audio/key).
- New `settings.ai.dictation` workspace flag (DTO + service + audit).
Frontend:
- Wire up the Voice/STT settings card (model/base URL/key) and the
Voice-dictation toggle.
- New `features/dictation`: useDictation (MediaRecorder state machine),
MicButton, transcribe service; integrated into the chat composer and a
new editor-toolbar dictation group, both gated by ai.dictation.
