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Speech recognition system performance degrades in noisy environments. If the acoustic models are built using features of clean utterances, the features of a noisy test utterance would be acoustically mismatched with the trained model. This…

Computation and Language · Computer Science 2015-07-16 D. S. Pavan Kumar

We introduce a new online adaptive filtering method called supervised multi-step adaptive filters (SMS-AF). Our method uses neural networks to control or optimize linear multi-delay or multi-channel frequency-domain filters and can flexibly…

Sound · Computer Science 2024-03-05 Jonah Casebeer , Nicholas J. Bryan , Paris Smaragdis

The reverberation time is one of the most important parameters used to characterize the acoustic property of an enclosure. In real-world scenarios, it is much more convenient to estimate the reverberation time blindly from recorded speech…

Sound · Computer Science 2021-12-10 Kaitong Zheng , Chengshi Zheng , Jinqiu Sang , Yulong Zhang , Xiaodong Li

Frequency modulation features capture the fine structure of speech formants that constitute beneficial and supplementary to the traditional energy-based cepstral features. Improvements have been demonstrated mainly in GMM-HMM systems for…

Sound · Computer Science 2019-09-04 Isidoros Rodomagoulakis , Petros Maragos

The Kalman filter is an established tool for the analysis of dynamic systems with normally distributed noise, and it has been successfully applied in numerous application areas. It provides sequentially calculated estimates of the system…

Systems and Control · Computer Science 2016-10-26 S. Eichstädt , N. Makarava , C. Elster

Reverberation results in reduced intelligibility for both normal and hearing-impaired listeners. This paper presents a novel psychoacoustic approach of dereverberation of a single speech source by recycling a pre-trained binaural anechoic…

Audio and Speech Processing · Electrical Eng. & Systems 2022-08-10 Sania Gul , Muhammad Salman Khan , Syed Waqar Shah , Ata Ur-Rehman

The Recursive KalmanNet, recently introduced by the authors, is a recurrent neural network guided by a Kalman filter, capable of estimating the state variables and error covariance of stochastic dynamic systems from noisy measurements,…

Signal Processing · Electrical Eng. & Systems 2025-08-26 Cyril Falcon , Hassan Mortada , Mathéo Clavaud , Jean-Philippe Michel

The objective of deep learning methods based on encoder-decoder architectures for music source separation is to approximate either ideal time-frequency masks or spectral representations of the target music source(s). The spectral…

Deep learning has been recently introduced for efficient acoustic howling suppression (AHS). However, the recurrent nature of howling creates a mismatch between offline training and streaming inference, limiting the quality of enhanced…

Audio and Speech Processing · Electrical Eng. & Systems 2023-08-21 Hao Zhang , Meng Yu , Yuzhong Wu , Tao Yu , Dong Yu

The augmented, iterated Kalman smoother is applied to system identification for inverse problems in evolutionary differential equations. In the augmented smoother, the unknown, time-dependent coefficients are included in the state vector,…

Methodology · Statistics 2019-11-19 Kurt S. Riedel

This article offers an elaborate description of a Kalman filter code employed in the active control system. Conventional active noise management methods usually employ an adaptive filter, such as the filtered reference least mean square…

Systems and Control · Electrical Eng. & Systems 2024-02-13 Guo Yu

There are many deterministic mathematical operations (e.g. compression, clipping, downsampling) that degrade speech quality considerably. In this paper we introduce a neural network architecture, based on a modification of the DiffWave…

Sound · Computer Science 2021-09-03 Jianwei Zhang , Suren Jayasuriya , Visar Berisha

This letter proposes a new time domain absorption approach designed to reduce masking components of speech signals under noisy-reverberant conditions. In this method, the non-stationarity of corrupted signal segments is used to detect…

Audio and Speech Processing · Electrical Eng. & Systems 2020-02-19 G. Zucatelli , R. Coelho

Adaptive algorithm based on multi-channel linear prediction is an effective dereverberation method balancing well between the attenuation of the long-term reverberation and the dereverberated speech quality. However, the abrupt change of…

Audio and Speech Processing · Electrical Eng. & Systems 2018-08-24 Teng Xiang , Jing Lu , Kai Chen

Optimization is often cast as a deterministic problem, where the solution is found through some iterative procedure such as gradient descent. However, when training neural networks the loss function changes over (iteration) time due to the…

Machine Learning · Computer Science 2025-03-25 Aram Davtyan , Sepehr Sameni , Llukman Cerkezi , Givi Meishvilli , Adam Bielski , Paolo Favaro

Automatic meeting analysis is an essential fundamental technology required to let, e.g. smart devices follow and respond to our conversations. To achieve an optimal automatic meeting analysis, we previously proposed an all-neural approach…

Audio and Speech Processing · Electrical Eng. & Systems 2020-03-10 Keisuke Kinoshita , Marc Delcroix , Shoko Araki , Tomohiro Nakatani

In realistic environments, speech is usually interfered by various noise and reverberation, which dramatically degrades the performance of automatic speech recognition (ASR) systems. To alleviate this issue, the commonest way is to use a…

Sound · Computer Science 2018-05-04 Bin Liu , Shuai Nie , Yaping Zhang , Dengfeng Ke , Shan Liang , Wenju Liu1

This paper investigates the use of extended Kalman filtering to train recurrent neural networks with rather general convex loss functions and regularization terms on the network parameters, including $\ell_1$-regularization. We show that…

Machine Learning · Computer Science 2022-11-03 Alberto Bemporad

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

Reverberation is damaging to both the quality and the intelligibility of a speech signal. We propose a novel single-channel method of dereverberation based on a linear filter in the Short Time Fourier Transform domain. Each enhanced frame…

Sound · Computer Science 2015-09-25 Richard Stanton , Mike Brookes