Streaming Speech-to-Text Translation with a SpeechLLM

· Source: Computation and Language · Field: Technology & Digital — Artificial Intelligence & Machine Learning · Depth: Expert, quick

Summary

A new LLM-based architecture has been developed for real streaming speech-to-text translation, addressing the limitations of existing SpeechLLM systems that suffer from high latency. Traditional systems typically use separate modules for speech recognition and text-to-text translation, leading to cascaded errors and an inability to leverage paralinguistic information. This proposed SpeechLLM learns to determine when it has processed sufficient audio to emit output tokens, rather than waiting for a complete utterance or outputting at fixed intervals. The system is trained using automatic alignments between input speech and output text. Evaluations across various language pairs demonstrate that this architecture achieves translation quality comparable to non-streaming baselines, while significantly reducing latency to just 1-2 seconds.

Key takeaway

For research scientists developing real-time communication systems, this streaming SpeechLLM architecture offers a viable path to significantly reduce translation latency to 1-2 seconds without sacrificing quality. You should explore integrating dynamic token emission logic into your LLM-based translation models to enable more responsive and natural user experiences in live applications.

Key insights

A SpeechLLM can achieve real-time streaming speech-to-text translation with low latency and high quality.

Principles

Method

An LLM learns to decide when to emit output tokens based on sufficient audio input, trained with automatic speech-text alignments.

In practice

Topics

Best for: Research Scientist, AI Scientist, NLP Engineer

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Editorial summary, takeaway, and curation by AIssential. Original article published by Computation and Language.