Minimum Latency Training of Sequence Transducers for Streaming End-to-End Speech Recognition
Yusuke Shinohara, Shinji Watanabe

TL;DR
This paper introduces a novel training method for sequence transducers in streaming speech recognition that explicitly models and minimizes latency, achieving significant latency reduction with minimal accuracy loss.
Contribution
The paper proposes a new training approach that explicitly incorporates latency modeling into sequence transducer training, improving latency-accuracy trade-offs.
Findings
Reduces latency from 220 ms to 27 ms with only 0.7% WER increase
Outperforms FastEmit and alignment-restricted training in latency reduction
Effective latency reduction demonstrated on WSJ dataset
Abstract
Sequence transducers, such as the RNN-T and the Conformer-T, are one of the most promising models of end-to-end speech recognition, especially in streaming scenarios where both latency and accuracy are important. Although various methods, such as alignment-restricted training and FastEmit, have been studied to reduce the latency, latency reduction is often accompanied with a significant degradation in accuracy. We argue that this suboptimal performance might be caused because none of the prior methods explicitly model and reduce the latency. In this paper, we propose a new training method to explicitly model and reduce the latency of sequence transducer models. First, we define the expected latency at each diagonal line on the lattice, and show that its gradient can be computed efficiently within the forward-backward algorithm. Then we augment the transducer loss with this expected…
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Taxonomy
TopicsSpeech Recognition and Synthesis · Speech and Audio Processing · Music and Audio Processing
MethodsNone
