gohanjanyan/speaker-diarization
Streaming Speaker Diarization
Real-time speaker diarization system developed as part of my Master's thesis at Yerevan State University (Applied Statistics & Data Science, 2026).
What it does
Upload or select a WAV audio file the system detects who speaks when, marks speaker change points, displays a speaker timeline, and shows a transcript with speaker labels.
Architecture
This system is designed for streaming, low-latency inference not offline batch processing. The key design choices:
- Frozen Emformer RNN-T encoder (pretrained ASR backbone) provides rich acoustic representations without retraining
- Custom SCChangeHead (~148K trainable parameters) detects speaker change boundaries in the streaming embedding sequence
- TitaNet-based speaker embeddings used for speaker identity verification via SCProjectionHead
- Trained with custom loss functions including Gaussian-smoothed cross-entropy (σ=2.5–3.0), InfoNCE, and clean suppression loss
The frozen encoder means the system is 1–2 orders of magnitude lighter in trainable parameters compared to comparable streaming diarization systems.
Results
Evaluated on a custom A–B–A protocol using LibriSpeech:
RTF < 1.0 means the system processes audio faster than real time suitable for live meeting transcription and audio intelligence applications.
How to use the demo
- Select a WAV file from the list
- Click Analyze
- View the speaker timeline and labeled transcript
About
Greta Ohanjanyan M.Sc. Applied Statistics & Data Science — Yerevan State University 📧 greta.ohanjanyan2001@gmail.com 🤗 huggingface.co/gohanjanyan
