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In this paper, we present a novel two-pass approach to unify streaming and non-streaming end-to-end (E2E) speech recognition in a single model. Our model adopts the hybrid CTC/attention architecture, in which the conformer layers in the…

声音 · 计算机科学 2021-12-30 Binbin Zhang , Di Wu , Zhuoyuan Yao , Xiong Wang , Fan Yu , Chao Yang , Liyong Guo , Yaguang Hu , Lei Xie , Xin Lei

Self-supervised pretrained models exhibit competitive performance in automatic speech recognition on finetuning, even with limited in-domain supervised data. However, popular pretrained models are not suitable for streaming ASR because they…

History and future contextual information are known to be important for accurate acoustic modeling. However, acquiring future context brings latency for streaming ASR. In this paper, we propose a new framework - Chunking, Simulating Future…

音频与语音处理 · 电气工程与系统科学 2022-08-03 Keyu An , Huahuan Zheng , Zhijian Ou , Hongyu Xiang , Ke Ding , Guanglu Wan

In this work we propose an inference technique, asynchronous revision, to unify streaming and non-streaming speech recognition models. Specifically, we achieve dynamic latency with only one model by using arbitrary right context during…

音频与语音处理 · 电气工程与系统科学 2020-11-04 Mingkun Huang , Meng Cai , Jun Zhang , Yang Zhang , Yongbin You , Yi He , Zejun Ma

Recent studies of streaming automatic speech recognition (ASR) recurrent neural network transducer (RNN-T)-based systems have fed the encoder with past contextual information in order to improve its word error rate (WER) performance. In…

音频与语音处理 · 电气工程与系统科学 2022-10-31 Alejandro Gomez-Alanis , Lukas Drude , Andreas Schwarz , Rupak Vignesh Swaminathan , Simon Wiesler

Distant-microphone meeting transcription is a challenging task. State-of-the-art end-to-end speaker-attributed automatic speech recognition (SA-ASR) architectures lack a multichannel noise and reverberation reduction front-end, which limits…

计算与语言 · 计算机科学 2025-07-09 Can Cui , Imran Ahamad Sheikh , Mostafa Sadeghi , Emmanuel Vincent

Research in auditory, visual, and audiovisual speech recognition (ASR, VSR, and AVSR, respectively) has traditionally been conducted independently. Even recent self-supervised studies addressing two or all three tasks simultaneously tend to…

计算机视觉与模式识别 · 计算机科学 2024-11-05 Alexandros Haliassos , Rodrigo Mira , Honglie Chen , Zoe Landgraf , Stavros Petridis , Maja Pantic

Encoder-decoder based sequence-to-sequence models have demonstrated state-of-the-art results in end-to-end automatic speech recognition (ASR). Recently, the transformer architecture, which uses self-attention to model temporal context…

声音 · 计算机科学 2020-07-02 Niko Moritz , Takaaki Hori , Jonathan Le Roux

We present a training scheme for streaming automatic speech recognition (ASR) based on recurrent neural network transducers (RNN-T) which allows the encoder network to learn to exploit context audio from a stream, using segmented or…

音频与语音处理 · 电气工程与系统科学 2021-06-16 Andreas Schwarz , Ilya Sklyar , Simon Wiesler

Streaming end-to-end automatic speech recognition (ASR) systems are widely used in everyday applications that require transcribing speech to text in real-time. Their minimal latency makes them suitable for such tasks. Unlike their…

计算与语言 · 计算机科学 2021-04-30 Thibault Doutre , Wei Han , Chung-Cheng Chiu , Ruoming Pang , Olivier Siohan , Liangliang Cao

The unified streaming and non-streaming speech recognition model has achieved great success due to its comprehensive capabilities. In this paper, we propose to improve the accuracy of the unified model by bridging the inherent…

计算与语言 · 计算机科学 2023-06-02 Yuting Yang , Yuke Li , Binbin Du

Automatic Speech Recognition (ASR) using multiple microphone arrays has achieved great success in the far-field robustness. Taking advantage of all the information that each array shares and contributes is crucial in this task. Motivated by…

计算与语言 · 计算机科学 2019-02-20 Xiaofei Wang , Ruizhi Li , Sri Harish Mallid , Takaaki Hori , Shinji Watanabe , Hynek Hermansky

Many Automatic Speech Recognition (ASR) applications require streaming processing of the audio data. In streaming mode, ASR systems need to start transcribing the input stream before it is complete, i.e., the systems have to process a…

计算与语言 · 计算机科学 2026-03-13 Youness Dkhissi , Valentin Vielzeuf , Elys Allesiardo , Anthony Larcher

Conformer-based end-to-end models have become ubiquitous these days and are commonly used in both streaming and non-streaming automatic speech recognition (ASR). Techniques like dual-mode and dynamic chunk training helped unify streaming…

音频与语音处理 · 电气工程与系统科学 2024-03-05 Goeric Huybrechts , Srikanth Ronanki , Xilai Li , Hadis Nosrati , Sravan Bodapati , Katrin Kirchhoff

It is difficult for an E2E ASR system to recognize words such as entities appearing infrequently in the training data. A widely used method to mitigate this issue is feeding contextual information into the acoustic model. Previous works…

声音 · 计算机科学 2023-06-09 Zhanheng Yang , Sining Sun , Xiong Wang , Yike Zhang , Long Ma , Lei Xie

Chunk-based inference stands out as a popular approach in developing real-time streaming speech recognition, valued for its simplicity and efficiency. However, because it restricts the model's focus to only the history and current chunk…

声音 · 计算机科学 2025-02-24 Khanh Le , Duc Chau

Non-autoregressive (NAR) modeling has gained more and more attention in speech processing. With recent state-of-the-art attention-based automatic speech recognition (ASR) structure, NAR can realize promising real-time factor (RTF)…

音频与语音处理 · 电气工程与系统科学 2021-07-21 Tianzi Wang , Yuya Fujita , Xuankai Chang , Shinji Watanabe

This work introduces the Cleanformer, a streaming multichannel neural based enhancement frontend for automatic speech recognition (ASR). This model has a conformer-based architecture which takes as inputs a single channel each of raw and…

音频与语音处理 · 电气工程与系统科学 2023-05-05 Joseph Caroselli , Arun Narayanan , Nathan Howard , Tom O'Malley

There is often a trade-off between performance and latency in streaming automatic speech recognition (ASR). Traditional methods such as look-ahead and chunk-based methods, usually require information from future frames to advance…

音频与语音处理 · 电气工程与系统科学 2022-07-07 Zehan Li , Haoran Miao , Keqi Deng , Gaofeng Cheng , Sanli Tian , Ta Li , Yonghong Yan

Self-supervised learning (SSL) based discrete speech representations are highly compact and domain adaptable. In this paper, SSL discrete speech features extracted from WavLM models are used as additional cross-utterance acoustic context…

计算与语言 · 计算机科学 2025-06-11 Mingyu Cui , Yifan Yang , Jiajun Deng , Jiawen Kang , Shujie Hu , Tianzi Wang , Zhaoqing Li , Shiliang Zhang , Xie Chen , Xunying Liu