Back to Portfolio

Real-Time Speech Analysis Platform

Production-grade live audio transcription system with Whisper ASR, real-time WebSocket streaming, multi-layer noise filtering, and speaker diarization. Features browser audio capture, VAD filtering, and hallucination prevention.

Project Screenshot

Technologies Used

PythonFastAPIReactWhisperfaster-whisperWebSocketFFmpegVADReal-time

About This Project

Real-time audio transcription system built with faster-whisper ASR engine, designed for live call analysis and sales conversations. Implements sophisticated multi-layer noise filtering strategy including pre-transcription VAD (Voice Activity Detection), optimized Whisper parameters, post-transcription hallucination detection, and intelligent deduplication. Features browser audio capture supporting both tab audio and microphone input, WebSocket streaming with jitter buffering, audio windowing for optimal transcription accuracy, speaker diarization with role assignment (sales rep/prospect), and FFmpeg-based audio processing pipeline. Built with React frontend for real-time visualization, FastAPI backend with asyncio for concurrent processing, and comprehensive quality controls achieving 90%+ hallucination filtering. Production-ready with configurable thresholds, extensive logging, and support for both CPU and GPU acceleration.

Key Features

  • • Feature 1: Add your project features here
  • • Feature 2: Describe the main functionality
  • • Feature 3: Highlight what makes it special

Challenges & Solutions

Describe the challenges you faced during development and how you solved them. This helps showcase your problem-solving skills.