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Single-channel speech enhancement (SE) is an important task in speech processing. A widely used framework combines an analysis/synthesis filterbank with a mask prediction network, such as the Conv-TasNet architecture. In such systems, the…

音频与语音处理 · 电气工程与系统科学 2021-08-06 Yuma Koizumi , Shigeki Karita , Scott Wisdom , Hakan Erdogan , John R. Hershey , Llion Jones , Michiel Bacchiani

We explore options to use Transformer networks in neural transducer for end-to-end speech recognition. Transformer networks use self-attention for sequence modeling and comes with advantages in parallel computation and capturing contexts.…

音频与语音处理 · 电气工程与系统科学 2019-10-30 Ching-Feng Yeh , Jay Mahadeokar , Kaustubh Kalgaonkar , Yongqiang Wang , Duc Le , Mahaveer Jain , Kjell Schubert , Christian Fuegen , Michael L. Seltzer

Streaming speech enhancement is a crucial task for real-time applications such as online meetings, smart home appliances, and hearing aids. Deep neural network-based approaches achieve exceptional performance while demanding substantial…

音频与语音处理 · 电气工程与系统科学 2025-09-29 Sunghwan Ahn , Jinmo Han , Beom Jun Woo , Nam Soo Kim

Recently, fully recurrent neural network (RNN) based end-to-end models have been proven to be effective for multi-speaker speech recognition in both the single-channel and multi-channel scenarios. In this work, we explore the use of…

音频与语音处理 · 电气工程与系统科学 2020-02-14 Xuankai Chang , Wangyou Zhang , Yanmin Qian , Jonathan Le Roux , Shinji Watanabe

On-device end-to-end speech recognition poses a high requirement on model efficiency. Most prior works improve the efficiency by reducing model sizes. We propose to reduce the complexity of model architectures in addition to model sizes.…

计算与语言 · 计算机科学 2020-11-12 Peidong Wang , DeLiang Wang

With excellent generalization ability, self-supervised speech models have shown impressive performance on various downstream speech tasks in the pre-training and fine-tuning paradigm. However, as the growing size of pre-trained models,…

音频与语音处理 · 电气工程与系统科学 2024-03-04 Mufan Sang , John H. L. Hansen

This study addresses robust automatic speech recognition (ASR) by introducing a Conformer-based acoustic model. The proposed model builds on the wide residual bi-directional long short-term memory network (WRBN) with utterance-wise dropout…

声音 · 计算机科学 2022-10-21 Yufeng Yang , Peidong Wang , DeLiang Wang

The Conformer has become the most popular encoder model for automatic speech recognition (ASR). It adds convolution modules to a transformer to learn both local and global dependencies. In this work we describe a faster, more…

音频与语音处理 · 电气工程与系统科学 2024-04-11 Zengwei Yao , Liyong Guo , Xiaoyu Yang , Wei Kang , Fangjun Kuang , Yifan Yang , Zengrui Jin , Long Lin , Daniel Povey

In this paper, we propose an end-to-end speech recognition network based on Nvidia's previous QuartzNet model. We try to promote the model performance, and design three components: (1) Multi-Resolution Convolution Module, replaces the…

音频与语音处理 · 电气工程与系统科学 2020-11-30 Jian Luo , Jianzong Wang , Ning Cheng , Guilin Jiang , Jing Xiao

With the rapid development of speech conversion and speech synthesis algorithms, automatic speaker verification (ASV) systems are vulnerable to spoofing attacks. In recent years, researchers had proposed a number of anti-spoofing methods…

声音 · 计算机科学 2022-12-23 Qiaowei Ma , Jinghui Zhong , Yitao Yang , Weiheng Liu , Ying Gao , Wing W. Y. Ng

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

The recurrent neural network transducer (RNN-T) is a prominent streaming end-to-end (E2E) ASR technology. In RNN-T, the acoustic encoder commonly consists of stacks of LSTMs. Very recently, as an alternative to LSTM layers, the Conformer…

In this paper, we propose an efficient and accurate streaming speech recognition model based on the FastConformer architecture. We adapted the FastConformer architecture for streaming applications through: (1) constraining both the…

计算与语言 · 计算机科学 2024-05-06 Vahid Noroozi , Somshubra Majumdar , Ankur Kumar , Jagadeesh Balam , Boris Ginsburg

End-to-end learning models using raw waveforms as input have shown superior performances in many audio recognition tasks. However, most model architectures are based on convolutional neural networks (CNN) which were mainly developed for…

音频与语音处理 · 电气工程与系统科学 2022-09-20 Taejun Kim , Juhan Nam

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

With increasingly more powerful compute capabilities and resources in today's devices, traditionally compute-intensive automatic speech recognition (ASR) has been moving from the cloud to devices to better protect user privacy. However, it…

Recently, conformer-based end-to-end automatic speech recognition, which outperforms recurrent neural network based ones, has received much attention. Although the parallel computing of conformer is more efficient than recurrent neural…

声音 · 计算机科学 2021-07-26 Shengqiang Li , Menglong Xu , Xiao-Lei Zhang

Speech enhancement is a demanding task in automated speech processing pipelines, focusing on separating clean speech from noisy channels. Transformer based models have recently bested RNN and CNN models in speech enhancement, however at the…

声音 · 计算机科学 2023-08-07 Jinyu Long , Jetic Gū , Binhao Bai , Zhibo Yang , Ping Wei , Junli Li

Attention-based encoder-decoder, e.g. transformer and its variants, generates the output sequence in an autoregressive (AR) manner. Despite its superior performance, AR model is computationally inefficient as its generation requires as many…

音频与语音处理 · 电气工程与系统科学 2024-09-27 Keyu An , Zerui Li , Zhifu Gao , Shiliang Zhang

In speaker verification, ECAPA-TDNN has shown remarkable improvement by utilizing one-dimensional(1D) Res2Net block and squeeze-and-excitation(SE) module, along with multi-layer feature aggregation (MFA). Meanwhile, in vision tasks, ConvNet…

音频与语音处理 · 电气工程与系统科学 2026-04-01 Hyun-Jun Heo , Ui-Hyeop Shin , Ran Lee , YoungJu Cheon , Hyung-Min Park