Prathamesh Sarjerao Vaidya
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metadata
title: Multilingual Audio Intelligence System
emoji: 🎡
colorFrom: blue
colorTo: purple
sdk: docker
pinned: false
short_description: AI system for multilingual transcription and translation

🎡 Multilingual Audio Intelligence System

Multilingual Audio Intelligence System Banner

Overview

The Multilingual Audio Intelligence System is an advanced AI-powered platform that combines state-of-the-art speaker diarization, automatic speech recognition, and neural machine translation to deliver comprehensive audio analysis capabilities. This sophisticated system processes multilingual audio content, identifies individual speakers, transcribes speech with high accuracy, and provides intelligent translations across multiple languages, transforming raw audio into structured, actionable insights.

Features

Demo Mode with Professional Audio Files

  • Yuri Kizaki - Japanese Audio: Professional voice message about website communication
  • French Film Podcast: Discussion about movies including Social Network and Paranormal Activity
  • Smart demo file management with automatic download and preprocessing
  • Instant results with cached processing for blazing-fast demonstration

Enhanced User Interface

  • Audio Waveform Visualization: Real-time waveform display with HTML5 Canvas
  • Interactive Demo Selection: Beautiful cards for selecting demo audio files
  • Improved Transcript Display: Color-coded confidence levels and clear translation sections
  • Professional Audio Preview: Audio player with waveform visualization

Screenshots

🎬 Demo Banner

Demo Banner

πŸ“ Transcript with Translation

Transcript with Translation

πŸ“Š Visual Representation

Visual Output

🧠 Summary Output

Summary Output

Demo & Documentation

Installation and Quick Start

  1. Clone the Repository:

    git clone https://github.com/Prathameshv07/Multilingual-Audio-Intelligence-System.git
    cd Multilingual-Audio-Intelligence-System
    
  2. Create and Activate Conda Environment:

    conda create --name audio_challenge python=3.9
    conda activate audio_challenge
    
  3. Install Dependencies:

    pip install -r requirements.txt
    
  4. Configure Environment Variables:

    cp config.example.env .env
    # Edit .env file with your HUGGINGFACE_TOKEN for accessing gated models
    
  5. Preload AI Models (Recommended):

    python model_preloader.py
    
  6. Initialize Application:

    python run_fastapi.py
    

File Structure

Multilingual-Audio-Intelligence-System/
β”œβ”€β”€ web_app.py                      # FastAPI application with RESTful endpoints
β”œβ”€β”€ model_preloader.py              # Intelligent model loading with progress tracking
β”œβ”€β”€ run_fastapi.py                  # Application startup script with preloading
β”œβ”€β”€ src/
β”‚   β”œβ”€β”€ main.py                     # AudioIntelligencePipeline orchestrator
β”‚   β”œβ”€β”€ audio_processor.py          # Advanced audio preprocessing and normalization
β”‚   β”œβ”€β”€ speaker_diarizer.py         # pyannote.audio integration for speaker identification
β”‚   β”œβ”€β”€ speech_recognizer.py        # faster-whisper ASR with language detection
β”‚   β”œβ”€β”€ translator.py               # Neural machine translation with multiple models
β”‚   β”œβ”€β”€ output_formatter.py         # Multi-format result generation and export
β”‚   └── utils.py                    # Utility functions and performance monitoring
β”œβ”€β”€ templates/
β”‚   └── index.html                  # Responsive web interface with home page
β”œβ”€β”€ static/                         # Static assets and client-side resources
β”œβ”€β”€ model_cache/                    # Intelligent model caching directory
β”œβ”€β”€ uploads/                        # User audio file storage
β”œβ”€β”€ outputs/                        # Generated results and downloads
β”œβ”€β”€ requirements.txt                # Comprehensive dependency specification
β”œβ”€β”€ Dockerfile                      # Production-ready containerization
└── config.example.env              # Environment configuration template

Configuration

Environment Variables

Create a .env file:

HUGGINGFACE_TOKEN=hf_your_token_here  # Optional, for gated models

Model Configuration

  • Whisper Model: tiny/small/medium/large
  • Target Language: en/es/fr/de/it/pt/zh/ja/ko/ar
  • Device: auto/cpu/cuda

Supported Audio Formats

  • WAV (recommended)
  • MP3
  • OGG
  • FLAC
  • M4A

Maximum file size: 100MB
Recommended duration: Under 30 minutes

Development

Local Development

python run_fastapi.py

Production Deployment

uvicorn web_app:app --host 0.0.0.0 --port 8000

Performance

  • Processing Speed: 2-14x real-time (depending on model size)
  • Memory Usage: Optimized with INT8 quantization
  • CPU Optimized: Works without GPU
  • Concurrent Processing: Async/await support

Troubleshooting

Common Issues

  1. Dependencies: Use requirements.txt for clean installation
  2. Memory: Use smaller models (tiny/small) for limited hardware
  3. Audio Format: Convert to WAV if other formats fail
  4. Port Conflicts: Change port in run_fastapi.py if 8000 is occupied

Error Resolution

  • Check logs in terminal output
  • Verify audio file format and size
  • Ensure all dependencies are installed
  • Check available system memory

Support

  • Documentation: Check /api/docs endpoint
  • System Info: Use the info button in the web interface
  • Logs: Monitor terminal output for detailed information

Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference