Hero Voice

Voice-to-markdown transcription server with real-time speech recognition, AI-powered text transformation, and live preview.

Features

• Real-time voice transcription - Stream audio from browser to server via WebSocket
• Voice Activity Detection - Silero VAD V5 neural network detects speech/silence transitions
• Automatic segmentation - Transcribes on natural pauses (350ms silence threshold)
• AI transcription - Uses Groq WhisperLargeV3Turbo with automatic failover
• Live markdown preview - Split-screen editor with real-time rendered HTML
• Text transformations - 14 built-in AI transformation styles:
  • spellcheck - Grammar and spelling correction
  • specs - Technical specifications
  • code - Software architecture documentation
  • docs - User-friendly documentation
  • legal - Legal document formatting
  • story - Creative narrative
  • summary - Bullet-point summary
  • technical - Technical documentation
  • business - Business analysis
  • meeting - Meeting minutes
  • email - Professional email
  • Language translations: Dutch, French, Arabic
• Topic organization - Hierarchical folder structure for transcriptions
• Audio archival - Saves recordings as WAV and compressed OGG

Requirements

• Rust 1.92+
• Groq API key (required for transcription)
• Modern browser with Web Audio API and microphone support

Configuration

# Required
export GROQ_API_KEY=your-groq-api-key

# Optional fallback providers
export OPENROUTER_API_KEY=your-openrouter-key
export SAMBANOVA_API_KEY=your-sambanova-key

# Server configuration (optional)
export RUST_LOG=hero_voice=info  # Log level

Usage

make run

Services listen on Unix sockets only (no TCP). Use hero_proxy for external access.

Sockets

Service	Socket Path
Server (OpenRPC)	~/hero/var/sockets/hero_voice_server.sock
UI (HTTP + /rpc proxy)	~/hero/var/sockets/hero_voice_ui.sock

Architecture

Hero Voice follows the standard Hero three-crate model:

hero_voice/
├── crates/
│   ├── hero_voice/              # Core library (types, domain logic, audio, transcription)
│   ├── hero_voice_server/       # JSON-RPC 2.0 server over Unix socket
│   ├── hero_voice_sdk/          # Generated client SDK
│   ├── hero_voice_ui/           # Admin UI (Axum HTTP + /rpc proxy + WebSocket)
│   └── hero_voice_examples/     # Example programs using the SDK
├── schemas/voice/voice.oschema  # Domain schema (source of truth)
├── data/                        # Runtime data (OTOML storage, audio, transforms)
├── Cargo.toml
├── Makefile
└── buildenv.sh

Data flow

Browser (WebSocket)
    │
    ▼
hero_voice_ui (Unix socket)
    ├── /rpc endpoint → proxies JSON-RPC to hero_voice_server.sock
    ├── /mcp endpoint → MCP-to-OpenRPC translation
    ├── /ws  endpoint → WebSocket audio streaming
    └── /*   fallback → embedded static assets

hero_voice_server (Unix socket)
    ├── rpc.health   → {"status":"ok"}
    ├── rpc.discover → OpenRPC spec
    └── domain methods (folder.*, topic.*, voiceservice.*)

API

JSON-RPC Endpoint

All data operations use JSON-RPC 2.0 via the /rpc proxy on the UI socket.

Auto-generated CRUD (Topic and Folder root objects):

• topic.new, topic.get, topic.set, topic.delete, topic.list
• folder.new, folder.get, folder.set, folder.delete, folder.list

Custom service methods (VoiceService):

• voiceservice.create_topic / voiceservice.create_folder
• voiceservice.rename_topic / voiceservice.rename_folder
• voiceservice.move_topic / voiceservice.move_folder
• voiceservice.delete_topic / voiceservice.delete_folder
• voiceservice.save_content / voiceservice.transform_content
• voiceservice.register_audio / voiceservice.delete_audio
• voiceservice.reset_topic / voiceservice.get_audio_path

WebSocket

GET /ws - Audio streaming endpoint

Client to Server:

{ "type": "start", "topic": "optional-topic-sid", "audio_dir": "optional-dir" }
{ "type": "stop" }

Plus binary audio data (16-bit PCM, 16kHz, mono, little-endian)

Server to Client:

{ "type": "transcription", "text": "...", "is_final": true }
{ "type": "status", "message": "..." }
{ "type": "error", "message": "..." }

Static Files

• GET /files/audio/{filename} - Audio file downloads
• GET /files/transforms/{filename} - Transform file downloads

Audio Processing

• Sample rate: 16kHz (required for Silero VAD)
• Chunk size: 512 samples for VAD analysis
• Silence threshold: 350ms triggers transcription
• Speech threshold: 0.20 probability
• Maximum buffer: 30 seconds before forced transcription
• Compression: OGG Vorbis at quality 0.4 (~10% of WAV size)

Browser Support

• Chrome 120+
• Firefox 120+
• Safari 17+
• Edge 120+

Requires microphone permission and WebSocket support.

Embedding & CORS

Hero Voice allows iframe embedding (no X-Frame-Options restrictions), cross-origin API calls, and WebSocket connections from any origin.

License

Apache-2.0