WhisperRT -- Turning Whisper into a Causal Streaming Model
Tomer Krichli, Bhiksha Raj, Joseph Keshet

TL;DR
This paper transforms the Whisper ASR model into a low-latency streaming system by making the encoder causal and fine-tuning the alignment, achieving better performance with lower complexity.
Contribution
It introduces a method to adapt transformer-based ASR models for streaming by causal encoding and alignment fine-tuning, with an improved inference mechanism.
Findings
Fine-tuned causal models outperform non-fine-tuned streaming approaches.
The approach achieves low latency (<300 ms) with lower complexity.
The method is validated through experiments on real-time audio chunks.
Abstract
Automatic Speech Recognition (ASR) has seen remarkable progress, with models like OpenAI Whisper and NVIDIA Canary achieving state-of-the-art (SOTA) performance in offline transcription. However, these models are not designed for streaming (online or real-time) transcription, due to limitations in their architecture and training methodology. We propose a method to turn the transformer encoder-decoder model into a low-latency streaming model. The encoder is made causal to process audio incrementally, while the decoder conditions on partial encoder states to generate tokens aligned with the available temporal context. This requires explicit synchronization between encoded input frames and token emissions. Since tokens are produced only after sufficient acoustic evidence is observed, an inherent latency arises, necessitating fine-tuning of the encoder-decoder alignment mechanism. We…
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