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Acoustic Echo Cancellation (AEC) is essential for accurate recognition of queries spoken to a smart speaker that is playing out audio. Previous work has shown that a neural AEC model operating on log-mel spectral features (denoted "logmel"…

音频与语音处理 · 电气工程与系统科学 2022-05-10 Sankaran Panchapagesan , Arun Narayanan , Turaj Zakizadeh Shabestary , Shuai Shao , Nathan Howard , Alex Park , James Walker , Alexander Gruenstein

Recently, the end-to-end approach has proven its efficacy in monaural multi-speaker speech recognition. However, high word error rates (WERs) still prevent these systems from being used in practical applications. On the other hand, the…

音频与语音处理 · 电气工程与系统科学 2019-10-17 Xuankai Chang , Wangyou Zhang , Yanmin Qian , Jonathan Le Roux , Shinji Watanabe

Combination approaches for speech recognition (ASR) systems cover structured sentence-level or word-based merging techniques as well as combination of model scores during beam search. In this work, we compare model combination across…

声音 · 计算机科学 2025-08-14 Noureldin Bayoumi , Robin Schmitt , Tina Raissi , Albert Zeyer , Ralf Schlüter , Hermann Ney

Connectionist temporal classification (CTC) and attention-based encoder decoder (AED) joint training has been widely applied in automatic speech recognition (ASR). Unlike most hybrid models that separately calculate the CTC and AED losses,…

计算与语言 · 计算机科学 2023-08-17 Daobin Zhu , Xiangdong Su , Hongbin Zhang

Self-supervised learning (SSL) methods which learn representations of data without explicit supervision have gained popularity in speech-processing tasks, particularly for single-talker applications. However, these models often have…

音频与语音处理 · 电气工程与系统科学 2022-11-02 Zili Huang , Desh Raj , Paola García , Sanjeev Khudanpur

We present a Conformer-based end-to-end neural diarization (EEND) model that uses both acoustic input and features derived from an automatic speech recognition (ASR) model. Two categories of features are explored: features derived directly…

计算与语言 · 计算机科学 2022-07-13 Aparna Khare , Eunjung Han , Yuguang Yang , Andreas Stolcke

This paper proposes a new approach to perform unsupervised fine-tuning and self-training using unlabeled speech data for recurrent neural network (RNN)-Transducer (RNN-T) end-to-end (E2E) automatic speech recognition (ASR) systems.…

计算与语言 · 计算机科学 2022-08-01 Cong-Thanh Do , Mohan Li , Rama Doddipatla

Convolutions have become essential in state-of-the-art end-to-end Automatic Speech Recognition~(ASR) systems due to their efficient modelling of local context. Notably, its use in Conformers has led to superior performance compared to…

计算与语言 · 计算机科学 2024-07-25 Darshan Prabhu , Yifan Peng , Preethi Jyothi , Shinji Watanabe

Training large foundation models using self-supervised objectives on unlabeled data, followed by fine-tuning on downstream tasks, has emerged as a standard procedure. Unfortunately, the efficacy of this approach is often constrained by both…

In Automatic Speech Recognition (ASR) systems, a recurring obstacle is the generation of narrowly focused output distributions. This phenomenon emerges as a side effect of Connectionist Temporal Classification (CTC), a robust sequence…

音频与语音处理 · 电气工程与系统科学 2024-03-19 SooHwan Eom , Eunseop Yoon , Hee Suk Yoon , Chanwoo Kim , Mark Hasegawa-Johnson , Chang D. Yoo

In this paper, we introduce a speech-conditioned Large Language Model (LLM) integrated with a Mixture of Experts (MoE) based connector to address the challenge of Code-Switching (CS) in Automatic Speech Recognition (ASR). Specifically, we…

声音 · 计算机科学 2024-11-01 Fengrun Zhang , Wang Geng , Hukai Huang , Yahui Shan , Cheng Yi , He Qu

End-to-end speech recognition systems usually require huge amounts of labeling resource, while annotating the speech data is complicated and expensive. Active learning is the solution by selecting the most valuable samples for annotation.…

音频与语音处理 · 电气工程与系统科学 2021-07-12 Jian Luo , Jianzong Wang , Ning Cheng , Jing Xiao

Multi-channel inputs offer several advantages over single-channel, to improve the robustness of on-device speech recognition systems. Recent work on multi-channel transformer, has proposed a way to incorporate such inputs into end-to-end…

音频与语音处理 · 电气工程与系统科学 2021-08-31 Feng-Ju Chang , Martin Radfar , Athanasios Mouchtaris , Maurizio Omologo

This paper summarizes the JHU team's efforts in tracks 1 and 2 of the CHiME-6 challenge for distant multi-microphone conversational speech diarization and recognition in everyday home environments. We explore multi-array processing…

Proper nouns present a challenge for end-to-end (E2E) automatic speech recognition (ASR) systems in that a particular name may appear only rarely during training, and may have a pronunciation similar to that of a more common word. Unlike…

音频与语音处理 · 电气工程与系统科学 2020-05-21 Cal Peyser , Tara N. Sainath , Golan Pundak

To realize robust end-to-end Automatic Speech Recognition(E2E ASR) under radio communication condition, we propose a multitask-based method to joint train a Speech Enhancement (SE) module as the front-end and an E2E ASR model as the…

音频与语音处理 · 电气工程与系统科学 2021-07-23 Duo Ma , Nana Hou , Van Tung Pham , Haihua Xu , Eng Siong Chng

Current Multimodal Large Language Models (MLLMs) rely on centralized architectures and often suffer from poor alignment between the input task and their fixed visual encoding modules, which limits performance on diverse and dynamic visual…

网络与互联网体系结构 · 计算机科学 2025-08-05 Yongjie Zeng , Hongyang Du

A deep neural network (DNN)-based speech enhancement (SE) aiming to maximize the performance of an automatic speech recognition (ASR) system is proposed in this paper. In order to optimize the DNN-based SE model in terms of the character…

音频与语音处理 · 电气工程与系统科学 2022-02-23 Ryosuke Sawata , Yosuke Kashiwagi , Shusuke Takahashi

State-of-the-art large-scale universal speech models (USMs) show a decent automatic speech recognition (ASR) performance across multiple domains and languages. However, it remains a challenge for these models to recognize overlapped speech,…

音频与语音处理 · 电气工程与系统科学 2023-05-31 Chenda Li , Yao Qian , Zhuo Chen , Naoyuki Kanda , Dongmei Wang , Takuya Yoshioka , Yanmin Qian , Michael Zeng

Array processing performance depends on the number of microphones available. Virtual microphone estimation (VME) has been proposed to increase the number of microphone signals artificially. Neural network-based VME (NN-VME) trains an NN…

音频与语音处理 · 电气工程与系统科学 2023-11-21 Hanako Segawa , Tsubasa Ochiai , Marc Delcroix , Tomohiro Nakatani , Rintaro Ikeshita , Shoko Araki , Takeshi Yamada , Shoji Makino