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In this paper, we present RT-GCC-NMF: a real-time (RT), two-channel blind speech enhancement algorithm that combines the non-negative matrix factorization (NMF) dictionary learning algorithm with the generalized cross-correlation (GCC)…

Audio and Speech Processing · Electrical Eng. & Systems 2019-04-08 Sean U. N. Wood , Jean Rouat

This paper introduces an explainable DNN-based beamformer with a postfilter (ExNet-BF+PF) for multichannel signal processing. Our approach combines the U-Net network with a beamformer structure to address this problem. The method involves a…

Audio and Speech Processing · Electrical Eng. & Systems 2024-11-19 Adi Cohen , Daniel Wong , Jung-Suk Lee , Sharon Gannot

Most deep learning-based multi-channel speech enhancement methods focus on designing a set of beamforming coefficients to directly filter the low signal-to-noise ratio signals received by microphones, which hinders the performance of these…

Sound · Computer Science 2022-02-08 Wenzhe Liu , Andong Li , Chengshi Zheng , Xiaodong Li

This work proposes a multichannel narrow-band speech separation network. In the short-time Fourier transform (STFT) domain, the proposed network processes each frequency independently, and all frequencies use a shared network. For each…

Sound · Computer Science 2022-12-06 Changsheng Quan , Xiaofei Li

Recent studies have demonstrated that incorporating auxiliary information, such as speaker voiceprint or visual cues, can substantially improve Speech Enhancement (SE) performance. However, single-channel methods often yield suboptimal…

Audio and Speech Processing · Electrical Eng. & Systems 2026-03-06 Chihyun Liu , Jiaxuan Fan , Mingtung Sun , Michael Anthony , Mingsian R. Bai , Yu Tsao

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…

Audio and Speech Processing · Electrical Eng. & Systems 2020-02-14 Xuankai Chang , Wangyou Zhang , Yanmin Qian , Jonathan Le Roux , Shinji Watanabe

Time-frequency (TF) domain dual-path models achieve high-fidelity speech separation. While some previous state-of-the-art (SoTA) models rely on RNNs, this reliance means they lack the parallelizability, scalability, and versatility of…

Audio and Speech Processing · Electrical Eng. & Systems 2024-08-08 Kohei Saijo , Gordon Wichern , François G. Germain , Zexu Pan , Jonathan Le Roux

Spatial filters can exploit deep-learning-based speech enhancement models to increase their reliability in scenarios with multiple speech sources scenarios. To further improve speech quality, it is common to perform postfiltering on the…

Audio and Speech Processing · Electrical Eng. & Systems 2024-09-13 François Grondin , Caleb Rascón

We study the segmental recurrent neural network for end-to-end acoustic modelling. This model connects the segmental conditional random field (CRF) with a recurrent neural network (RNN) used for feature extraction. Compared to most previous…

Computation and Language · Computer Science 2016-06-21 Liang Lu , Lingpeng Kong , Chris Dyer , Noah A. Smith , Steve Renals

Time-domain training criteria have proven to be very effective for the separation of single-channel non-reverberant speech mixtures. Likewise, mask-based beamforming has shown impressive performance in multi-channel reverberant speech…

Most current speech enhancement models use spectrogram features that require an expensive transformation and result in phase information loss. Previous work has overcome these issues by using convolutional networks to learn long-range…

Audio and Speech Processing · Electrical Eng. & Systems 2019-04-17 Jalal Abdulbaqi , Yue Gu , Ivan Marsic

This paper proposes a deep neural network (DNN)-based multi-channel speech enhancement system in which a DNN is trained to maximize the quality of the enhanced time-domain signal. DNN-based multi-channel speech enhancement is often…

Audio and Speech Processing · Electrical Eng. & Systems 2020-02-17 Yoshiki Masuyama , Masahito Togami , Tatsuya Komatsu

Bridge models have been investigated in speech enhancement but are mostly single-task, with constrained general speech restoration (GSR) capability. In this work, we propose VoiceBridge, a one-step latent bridge model (LBM) for GSR, capable…

Sound · Computer Science 2026-03-11 Chi Zhang , Kaiwen Zheng , Zehua Chen , Jun Zhu

Single-channel, speaker-independent speech separation methods have recently seen great progress. However, the accuracy, latency, and computational cost of such methods remain insufficient. The majority of the previous methods have…

Sound · Computer Science 2019-05-16 Yi Luo , Nima Mesgarani

State-of-the-art neural network language models (NNLMs) represented by long short term memory recurrent neural networks (LSTM-RNNs) and Transformers are becoming highly complex. They are prone to overfitting and poor generalization when…

Computation and Language · Computer Science 2022-08-30 Boyang Xue , Shoukang Hu , Junhao Xu , Mengzhe Geng , Xunying Liu , Helen Meng

Monaural source separation is important for many real world applications. It is challenging because, with only a single channel of information available, without any constraints, an infinite number of solutions are possible. In this paper,…

Sound · Computer Science 2015-10-02 Po-Sen Huang , Minje Kim , Mark Hasegawa-Johnson , Paris Smaragdis

We propose a spatial diffuseness feature for deep neural network (DNN)-based automatic speech recognition to improve recognition accuracy in reverberant and noisy environments. The feature is computed in real-time from multiple microphone…

Computation and Language · Computer Science 2015-09-02 Andreas Schwarz , Christian Huemmer , Roland Maas , Walter Kellermann

Deep neural network (DNN) based end-to-end optimization in the complex time-frequency (T-F) domain or time domain has shown considerable potential in monaural speech separation. Many recent studies optimize loss functions defined solely in…

Sound · Computer Science 2022-01-05 Zhong-Qiu Wang , Gordon Wichern , Jonathan Le Roux

Multichannel linear filters, such as the Multichannel Wiener Filter (MWF) and the Generalized Eigenvalue (GEV) beamformer are popular signal processing techniques which can improve speech recognition performance. In this paper, we present…

Sound · Computer Science 2017-11-16 Ziteng Wang , Emmanuel Vincent , Romain Serizel , Yonghong Yan

We present a transformer-based speech-declipping model that effectively recovers clipped signals across a wide range of input signal-to-distortion ratios (SDRs). While recent time-domain deep neural network (DNN)-based declippers have…

Audio and Speech Processing · Electrical Eng. & Systems 2024-09-20 Younghoo Kwon , Jung-Woo Choi