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local-transcription/config/default_config.yaml

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user:
name: "User"
id: ""
audio:
input_device: "default"
sample_rate: 16000
transcription:
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
# RealtimeSTT model settings
model: "base.en" # Options: tiny, tiny.en, base, base.en, small, small.en, medium, medium.en, large-v1, large-v2, large-v3
device: "auto" # auto, cuda, cpu
language: "en"
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
compute_type: "default" # default, int8, float16, float32
# Realtime preview settings (optional faster preview before final transcription)
enable_realtime_transcription: false
realtime_model: "tiny.en" # Faster model for instant preview
realtime_processing_pause: 0.1 # Seconds between preview updates (lower = more responsive, default 0.1)
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
# VAD (Voice Activity Detection) settings
silero_sensitivity: 0.4 # 0.0-1.0, lower = more sensitive (detects more speech)
silero_use_onnx: true # Use ONNX for 2-3x faster VAD with lower CPU usage
webrtc_sensitivity: 3 # 0-3, lower = more sensitive
# Post-processing settings
post_speech_silence_duration: 0.3 # Seconds of silence before finalizing transcription
min_length_of_recording: 0.5 # Minimum recording length in seconds
min_gap_between_recordings: 0 # Minimum gap between recordings in seconds
pre_recording_buffer_duration: 0.2 # Buffer before speech starts (prevents cut-off words)
# Transcription quality settings
beam_size: 5 # Higher = better quality but slower (1-10)
initial_prompt: "" # Optional prompt to guide transcription style
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
# Performance settings
no_log_file: true # Disable RealtimeSTT logging
# Fast speaker mode - for speakers who talk quickly without pauses
# Reduces silence detection thresholds for more frequent transcription outputs
continuous_mode: false
server_sync:
enabled: false
url: "http://localhost:3000/api/send"
room: "default"
passphrase: ""
# Font settings are now in the display section (shared for local and server sync)
display:
show_timestamps: true
max_lines: 100
# Font settings (used for both local display and server sync)
font_source: "System Font" # Options: System Font, Web-Safe, Google Font, Custom File
font_family: "Courier" # System font name (local only, won't work with server sync)
websafe_font: "Arial" # Web-safe font name
google_font: "Roboto" # Google Font name
custom_font_file: "" # Path to custom font file (.ttf, .otf, .woff, .woff2)
font_size: 12
theme: "dark"
fade_after_seconds: 10 # Time before transcriptions fade out (0 = never fade)
# Color settings (used for both local display and server sync)
user_color: "#4CAF50" # User's name color (default green)
text_color: "#FFFFFF" # Text/font color (default white)
background_color: "#000000B3" # Background color with alpha (default semi-transparent black)
web_server:
port: 8080
host: "127.0.0.1"
remote_processing:
enabled: false # Enable remote transcription offloading
server_url: "" # WebSocket URL of remote transcription service (e.g., ws://your-server:8765/ws/transcribe)
api_key: "" # API key for authentication
fallback_to_local: true # Fall back to local processing if remote fails
updates:
auto_check: true # Check for updates on startup
gitea_url: "https://repo.anhonesthost.net" # Base URL of Gitea server
owner: "streamer-tools" # Repository owner/organization name
repo: "local-transcription" # Repository name
skipped_versions: [] # List of versions the user chose to skip
last_check: "" # ISO timestamp of last update check
check_interval_hours: 24 # Hours between automatic update checks