neuralvoiceGPU / IMPLEMENTATION_NOTES.md
Shantika
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Implementation Notes

Architecture Overview

The STT system is built in 5 progressive steps, each adding functionality on top of the previous:

  1. Step 1: Basic offline transcription (Whisper/Vosk)
  2. Step 2: HTTP API for file uploads
  3. Step 3: WebSocket streaming for real-time audio
  4. Step 4: Telephony audio format support (Twilio/Exotel)
  5. Step 5: Production-ready with stability features

Key Components

Audio Processing

  • TelephonyAudioConverter: Handles format conversion
    • Twilio: 8kHz μ-law → 16kHz PCM
    • Exotel: 8kHz PCM → 16kHz PCM
    • Uses scipy.signal.resample for sample rate conversion

Voice Activity Detection (VAD)

  • Simple energy-based VAD in Step 5
  • Threshold: 0.01 (configurable)
  • Frame-based analysis (25ms frames)
  • Detects speech vs silence

Audio Buffering

  • AudioBuffer: Accumulates audio chunks
  • Configurable chunk duration (default: 1.0s)
  • Minimum interval between transcriptions (0.5s)
  • Handles silence timeouts (3.0s)

Duplicate Prevention

  • Compares new transcriptions with previous
  • Prevents sending identical text multiple times
  • Simple substring matching (can be enhanced)

Things to Consider

Performance

  1. Model Loading: Whisper models are loaded per connection (lazy loading)

    • Consider model caching/pooling for production
    • Larger models (medium/large) are more accurate but slower
  2. Chunk Size: Balance between latency and accuracy

    • Smaller chunks = lower latency but less context
    • Larger chunks = better accuracy but higher latency
  3. Concurrent Connections: Each connection loads its own model

    • Consider shared model instances for multiple connections
    • Monitor memory usage with many concurrent calls

Audio Quality

  1. Sample Rate: Whisper works best with 16kHz

    • Telephony audio (8kHz) must be upsampled
    • Quality may be reduced compared to native 16kHz
  2. Noise: Telephony audio often has background noise

    • Consider noise reduction preprocessing
    • VAD helps filter silence but not noise
  3. Format Conversion: μ-law to PCM conversion may introduce artifacts

    • Test with real telephony audio
    • Consider alternative conversion methods if quality is poor

Stability & Reliability

  1. Disconnections: Handled gracefully in Step 5

    • Final transcription on remaining buffer
    • Session cleanup on disconnect
  2. Error Handling: Comprehensive error catching

    • Logs errors per call
    • Continues processing on individual failures
  3. Logging: Per-call logging in Step 5

    • Logs stored in logs/stt.log
    • Includes call_id for tracking

Scaling Considerations

  1. Model Memory: Whisper models are large (base ~150MB, large ~3GB)

    • Consider GPU acceleration for faster inference
    • Model quantization for reduced memory
  2. API Rate Limiting: No rate limiting implemented

    • Add rate limiting for production
    • Consider request queuing
  3. Database: No persistent storage

    • Add database for call transcripts
    • Store session metadata
  4. Load Balancing: Single server implementation

    • Consider multiple workers/instances
    • Use message queue for audio processing

Security

  1. Authentication: No authentication implemented

    • Add API keys/tokens
    • WebSocket authentication
  2. Input Validation: Basic validation

    • Validate audio format/size
    • Rate limit per client
  3. Data Privacy: Transcripts logged

    • Consider encryption for sensitive data
    • Implement data retention policies

Testing Recommendations

  1. Unit Tests: Test audio conversion functions
  2. Integration Tests: Test WebSocket streaming with real audio
  3. Load Tests: Test with multiple concurrent connections
  4. Telephony Tests: Test with actual Twilio/Exotel audio streams

Future Enhancements

  1. Better VAD: Use more sophisticated VAD (e.g., WebRTC VAD)
  2. Streaming Model: Use streaming-capable models for lower latency
  3. Language Detection: Auto-detect language
  4. Speaker Diarization: Identify different speakers
  5. Punctuation: Better punctuation in transcripts
  6. Timestamping: Word-level timestamps
  7. Confidence Scores: Return confidence scores per word