Publication | Closed Access
A Streaming On-Device End-To-End Model Surpassing Server-Side Conventional Model Quality and Latency
201
Citations
21
References
2020
Year
Unknown Venue
EngineeringComputer ArchitectureWord Error RateSpoken Language ProcessingData Streaming ArchitectureStreaming DataRecurrent Neural NetworkSpeech RecognitionNatural Language ProcessingAcoustic DiversityReal-time LanguageMachine TranslationHealth SciencesAdaptive Bitrate StreamingStreaming EngineComputer EngineeringMobile ComputingComputer ScienceLatency FrontMultimedia DeliveryDeep LearningSpeech CommunicationMulti-speaker Speech RecognitionCloud ComputingSpeech ProcessingSpeech InputSpeech PerceptionLinguistics
Thus far, end-to-end (E2E) models have not been shown to outperform state-of-the-art conventional models with respect to both quality, i.e., word error rate (WER), and latency, i.e., the time the hypothesis is finalized after the user stops speaking. In this paper, we develop a first-pass Recurrent Neural Network Transducer (RNN-T) model and a second-pass Listen, Attend, Spell (LAS) rescorer that surpasses a conventional model in both quality and latency. On the quality side, we incorporate a large number of utterances across varied domains [1] to increase acoustic diversity and the vocabulary seen by the model. We also train with accented English speech to make the model more robust to different pronunciations. In addition, given the increased amount of training data, we explore a varied learning rate schedule. On the latency front, we explore using the end-of-sentence decision emitted by the RNN-T model to close the microphone, and also introduce various optimizations to improve the speed of LAS rescoring. Overall, we find that RNN-T+LAS offers a better WER and latency tradeoff compared to a conventional model. For example, for the same latency, RNN-T+LAS obtains a 8% relative improvement in WER, while being more than 400-times smaller in model size.
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