early preview - current state still contains lots of bugs and unsolved edge cases

Implements a real-time, voice-based chat application where users can speak directly to an AI assistant and receive spoken responses, mimicking a natural conversation flow. It utilizes a client-server architecture with WebSockets for low-latency communication.

The system captures microphone audio from a web client, streams it to a Python backend server, transcribes the audio to text in real-time, processes the text using a Large Language Model (LLM), synthesizes the AI's text response back into audio, and streams the audio back to the client for playback. It features interruption handling, turn detection, and displays partial transcriptions/responses.

• Real-Time Voice Interaction: Speak naturally and get spoken responses from the AI.
• Client-Server Architecture: Web-based client connects to a powerful Python backend via WebSockets.
• Low Latency: Optimized for minimal delay using audio chunk streaming.
• Real-Time Transcription: Uses RealtimeSTT for fast and accurate speech-to-text conversion.
• Turn Detection: Employs a model (turndetect.py) to dynamically adjust silence thresholds for natural conversation flow.
• LLM Integration: Connects to LLMs (configurable, supports Ollama and potentially OpenAI via inference.py) for intelligent responses.
• Real-Time Text-to-Speech (TTS): Uses RealtimeTTS with various engine options (Kokoro, Coqui, Orpheus) to generate spoken audio (audio_out.py).
• Partial & Final Responses: Displays user transcriptions and AI responses as they are generated.
• Interruption Handling: Allows the user to interrupt the AI's response by speaking.
• Web-Based UI: Simple and clean chat interface using HTML, CSS, and JavaScript (static/).
• Audio Worklets: Efficient client-side audio processing for capture and playback.

• Backend: Python 3.x, FastAPI
• Frontend: HTML, CSS, JavaScript (Vanilla JS, Web Audio API with AudioWorklets)
• Communication: WebSockets
• Core AI/ML Libraries:
  • RealtimeSTT (Speech-to-Text)
  • RealtimeTTS (Text-to-Speech)
  • transformers (For turn detection model, LLM tokenization)
  • torch / torchaudio (Required by STT/TTS/Transformers)
  • ollama / openai (LLM Interaction)
• Audio Processing: numpy, scipy
• Environment: Virtual Environment (venv)

Prerequisites:

• Python 3.9 or higher recommended.
• Windows recommended.
• A CUDA-enabled STRONG GPU is highly recommended for faster STT/TTS performance (especially for Coqui TTS and larger Whisper models). The installation script assumes CUDA 12.4 (cu124). Adjust if necessary.
• (Optional) Ollama: If using the Ollama backend for the LLM, ensure it is installed and running. Pull the desired model (e.g., ollama pull hf.co/bartowski/huihui-ai_Mistral-Small-24B-Instruct-2501-abliterated-GGUF:Q4_K_M), set in handlerequests.py as MODEL parameter.
• (Optional) OpenAI API Key: If using the OpenAI backend, set the OPENAI_API_KEY environment variable or place it in a .env file.

Installation Steps:

1. Clone the repository:

   git clone <repository-url>
   cd <repository-directory>

2. Run the installation script (Windows): This script automates creating a virtual environment, upgrading pip, installing PyTorch with CUDA support, and installing all other dependencies from requirements.txt.

   install.bat

   (Note: This will open a new command prompt window within the activated virtual environment.)

3. Manual Installation (Linux/macOS or if .bat fails):

   # Create virtual environment
   python -m venv venv
   
   # Activate virtual environment
   # Linux/macOS:
   source venv/bin/activate
   # Windows:
   .\venv\Scripts\activate
   
   # Upgrade pip
   python -m pip install --upgrade pip
   
   # Navigate to the code directory
   cd code
   
   # Install PyTorch with CUDA (adjust index-url for your CUDA version or CPU)
   # Example for CUDA 12.4:
   pip install torch torchaudio --index-url https://download.pytorch.org/whl/cu124
   # Example for CPU only:
   # pip install torch torchaudio
   
   # Install other requirements
   pip install -r requirements.txt

1. Ensure your virtual environment is activated.

   # If not already active:
   # Linux/macOS: source ../venv/bin/activate
   # Windows: ..\venv\Scripts\activate

2. Make sure you are in the code directory.

3. Start the FastAPI server:

   python server.py

4. Access the client: Open your web browser and navigate to http://localhost:8000 (or http://<your-server-ip>:8000 if running on a different machine).

5. Grant microphone permissions when prompted by the browser.

6. Click the "Start" button to begin the voice chat. Click "Stop" to end the session. Use "Reset" to clear the conversation history on both client and server.

Several aspects of the application can be configured by modifying the Python source files:

• TTS Engine (server.py, audio_out.py): Change START_ENGINE in server.py to "coqui", "kokoro", or "orpheus". Configure engine-specific settings (voice, speed, etc.) within AudioOutProcessor.__init__ in audio_out.py. Note: Coqui requires a reference_audio.wav file.
• LLM Model & Backend (handlerequests.py, inference.py):
  • Set the desired MODEL (e.g., Ollama model name or HF path) and TOKENIZER_MODEL in handlerequests.py.
  • Modify DEFAULT_BACKEND and model names (OPENAI_MODEL, OLLAMA_MODEL) in inference.py or use environment variables (LLM_BACKEND, OPENAI_API_KEY).
  • Adjust system prompts (fast_answer_system_prompt, orpheus_prompt_addon) in handlerequests.py.
• STT Settings (transcribe.py): Modify recorder_cfg within TranscriptionProcessor._create_recorder to change Whisper model size, language, sensitivities, silence durations, etc.
• Turn Detection (turndetect.py): Adjust pause duration constants (ellipsis_pause, punctuation_pause, etc.) for different speaking styles.
• SSL (server.py): Set USE_SSL = True and provide certificate/key files if HTTPS is required.

Contributions are welcome! Please feel free to submit pull requests or open issues for bugs, feature requests, or improvements.

My own Codebase is MIT. Please respect the license systems of TTS providers.