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Hands-free speech recaster powered by WebRTC, Whisper-Turbo, and GPT-OSS-20b for real-time natural English expressions.

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Tutur: Real-Time AI Speech Recasting Companion

Tutur is a privacy-first, zero-latency AI speech recasting web application designed for non-native English learners. Built to help users practice spoken English without fear or judgment, Tutur listens to incoming real-time audio streams, transcribes spoken phrases, and instantly reformulates them into natural, fluent native expressions while preserving the user's original intent and tone.

Powered by a FastAPI backend integrated with FastRTC for WebRTC audio handling, the system utilizes Silero VAD for intelligent pause detection. Spoken audio is processed on-the-fly via Groq's high-speed API—leveraging Whisper Large v3 Turbo for Speech-to-Text and GPT OSS 20B for real-time sentence recasting. The frontend is built as a reactive Single Page Application (SPA) using Alpine.js and Tailwind CSS, communicating over WebRTC data channels for instant live feed updates.

Application Screenshots

Screen Description Preview
Idle Companion Workspace The minimalist initial state displaying session status indicators, microphone trigger, and usage guidance. Idle Workspace
Live Recast Session Real-time interactive session feed showing speech-to-text transcriptions ("You") alongside native reformulations ("Versi natural") as you speak. Live Recast Session

System Features

  • Real-Time WebRTC Audio Streaming: High-throughput, send-only audio streaming over WebRTC paired with server-side Silero Voice Activity Detection (VAD) that triggers recasting upon natural pauses.
  • Dual-Stage Groq AI Pipeline: Ultra-fast Speech-to-Text using whisper-large-v3-turbo combined with openai/gpt-oss-20b for instant grammar, tense, and collocation reformulations.
  • Dynamic ICE/TURN Credential Management: Automated background renewal and serving of short-lived TURN/STUN credentials to guarantee seamless WebRTC NAT/firewall traversal.
  • Zero-Storage & Privacy-First: Audio buffer arrays are converted to PCM in-memory on demand; no raw audio files, transcriptions, or user sessions are ever stored on disk.
  • Reactive Stream UI: Lightweight Alpine.js SPA with auto-scrolling conversation feed, connection state management, and real-time visual distinction for already-natural phrasing.

Technical Specifications

  • Backend Framework: FastAPI (Python)
  • Real-Time Streaming Engine: FastRTC (fastrtc) over WebRTC
  • Voice Activity Detection: Silero VAD (SileroVadOptions & ReplyOnPause)
  • Speech-to-Text (STT): Whisper Large v3 Turbo (whisper-large-v3-turbo via Groq Async API)
  • Language Model (LLM): GPT OSS 20B (openai/gpt-oss-20b via Groq)
  • Network Traversal: Dynamic STUN/TURN Relay Server Management
  • Frontend Architecture: Single Page Application (Alpine.js / Tailwind CSS / FontAwesome)
  • Audio Processing Engine: NumPy, Wave, In-Memory BytesIO
  • License: MIT License

License

This project is licensed under the MIT License - see the LICENSE file for details.

About

Hands-free speech recaster powered by WebRTC, Whisper-Turbo, and GPT-OSS-20b for real-time natural English expressions.

Topics

Resources

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2 stars

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0 watching

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