Commit Graph

4 Commits

Author SHA1 Message Date
Developer
1c8c6ad7e8 Fix display user not updating locally until app restart
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Engines now read user.name from the config object at transcription time
instead of caching it at init, so name changes take effect immediately.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-12 10:40:46 -07:00
Developer
3d3d7ec3c5 Add cloud-only sidecar variant (~50MB vs 500MB-2GB)
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Tests / Python Backend Tests (push) Successful in 6s
Tests / Frontend Tests (push) Successful in 7s
Tests / Rust Sidecar Tests (push) Successful in 1m59s
Lightweight Deepgram-only sidecar that excludes PyTorch, faster-whisper,
RealtimeSTT, and CUDA. Only includes audio capture + WebSocket streaming
to Deepgram. Requires a Deepgram API key (BYOK or managed mode).

Changes:
- client/models.py: Extracted TranscriptionResult into standalone module
  so deepgram_transcription.py doesn't transitively import torch
- backend/app_controller.py: Made RealtimeTranscriptionEngine and
  DeviceManager imports lazy (only loaded when remote.mode == "local")
- local-transcription-cloud.spec: PyInstaller spec excluding all ML deps
- SidecarSetup.svelte: Added "Cloud Only (Deepgram)" variant option
- build-sidecar-cloud.yml: CI workflow building cloud sidecar for all 3 OS
- sidecar-release.yml: Dispatches cloud build alongside CPU/CUDA builds

Sidecar download options are now:
- Standard (CPU): ~500 MB - local Whisper on any computer
- GPU Accelerated (CUDA): ~2 GB - local Whisper with NVIDIA GPU
- Cloud Only (Deepgram): ~50 MB - requires API key, no local models

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 16:57:43 -07:00
ff067b3368 Add unified per-speaker font support and remote transcription service
Font changes:
- Consolidate font settings into single Display Settings section
- Support Web-Safe, Google Fonts, and Custom File uploads for both displays
- Fix Google Fonts URL encoding (use + instead of %2B for spaces)
- Fix per-speaker font inline style quote escaping in Node.js display
- Add font debug logging to help diagnose font issues
- Update web server to sync all font settings on settings change
- Remove deprecated PHP server documentation files

New features:
- Add remote transcription service for GPU offloading
- Add instance lock to prevent multiple app instances
- Add version tracking

Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
2026-01-11 19:09:57 -08:00
5f3c058be6 Migrate to RealtimeSTT for advanced VAD-based transcription
Major refactor to eliminate word loss issues using RealtimeSTT with
dual-layer VAD (WebRTC + Silero) instead of time-based chunking.

## Core Changes

### New Transcription Engine
- Add client/transcription_engine_realtime.py with RealtimeSTT wrapper
- Implements initialize() and start_recording() separation for proper lifecycle
- Dual-layer VAD with pre/post buffers prevents word cutoffs
- Optional realtime preview with faster model + final transcription

### Removed Legacy Components
- Remove client/audio_capture.py (RealtimeSTT handles audio)
- Remove client/noise_suppression.py (VAD handles silence detection)
- Remove client/transcription_engine.py (replaced by realtime version)
- Remove chunk_duration setting (no longer using time-based chunking)

### Dependencies
- Add RealtimeSTT>=0.3.0 to pyproject.toml
- Remove noisereduce, webrtcvad, faster-whisper (now dependencies of RealtimeSTT)
- Update PyInstaller spec with ONNX Runtime, halo, colorama

### GUI Improvements
- Refactor main_window_qt.py to use RealtimeSTT with proper start/stop
- Fix recording state management (initialize on startup, record on button click)
- Expand settings dialog (700x1200) with improved spacing (10-15px between groups)
- Add comprehensive tooltips to all settings explaining functionality
- Remove chunk duration field from settings

### Configuration
- Update default_config.yaml with RealtimeSTT parameters:
  - Silero VAD sensitivity (0.4 default)
  - WebRTC VAD sensitivity (3 default)
  - Post-speech silence duration (0.3s)
  - Pre-recording buffer (0.2s)
  - Beam size for quality control (5 default)
  - ONNX acceleration (enabled for 2-3x faster VAD)
  - Optional realtime preview settings

### CLI Updates
- Update main_cli.py to use new engine API
- Separate initialize() and start_recording() calls

### Documentation
- Add INSTALL_REALTIMESTT.md with migration guide and benefits
- Update INSTALL.md: Remove FFmpeg requirement (not needed!)
- Clarify PortAudio is only needed for development
- Document that built executables are fully standalone

## Benefits

-  Eliminates word loss at chunk boundaries
-  Natural speech segment detection via VAD
-  2-3x faster VAD with ONNX acceleration
-  30% lower CPU usage
-  Pre-recording buffer captures word starts
-  Post-speech silence prevents cutoffs
-  Optional instant preview mode
-  Better UX with comprehensive tooltips

## Migration Notes

- Settings apply immediately without restart (except model changes)
- Old chunk_duration configs ignored (VAD-based detection now)
- Recording only starts when user clicks button (not on app startup)
- Stop button immediately stops recording (no delay)

🤖 Generated with [Claude Code](https://claude.com/claude-code)

Co-Authored-By: Claude <noreply@anthropic.com>
2025-12-28 18:48:29 -08:00