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We propose an end-to-end joint optimization framework of a multi-channel neural speech extraction and deep acoustic model without mel-filterbank (FBANK) extraction for overlapped speech recognition. First, based on a multi-channel…

音频与语音处理 · 电气工程与系统科学 2019-10-31 Bo Wu , Meng Yu , Lianwu Chen , Chao Weng , Dan Su , Dong Yu

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…

声音 · 计算机科学 2019-09-04 Isidoros Rodomagoulakis , Petros Maragos

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…

声音 · 计算机科学 2017-11-16 Ziteng Wang , Emmanuel Vincent , Romain Serizel , Yonghong Yan

Recent single-channel speech enhancement methods based on deep neural networks (DNNs) have achieved remarkable results, but there are still generalization problems in real scenes. Like other data-driven methods, DNN-based speech enhancement…

音频与语音处理 · 电气工程与系统科学 2021-07-12 Lu Zhang , Mingjiang Wang , Andong Li , Zehua Zhang , Xuyi Zhuang

The development of neural vocoders (NVs) has resulted in the high-quality and fast generation of waveforms. However, conventional NVs target a single sampling rate and require re-training when applied to different sampling rates. A suitable…

音频与语音处理 · 电气工程与系统科学 2022-06-27 Kentaro Mitsui , Kei Sawada

Spectral subtraction, widely used for its simplicity, has been employed to address the Robot Ego Speech Filtering (RESF) problem for detecting speech contents of human interruption from robot's single-channel microphone recordings when it…

机器人学 · 计算机科学 2024-09-11 Yue Li , Koen V. Hindriks , Florian A. Kunneman

The time delay neural network (TDNN) represents one of the state-of-the-art of neural solutions to text-independent speaker verification. However, they require a large number of filters to capture the speaker characteristics at any local…

声音 · 计算机科学 2022-02-16 Tianchi Liu , Rohan Kumar Das , Kong Aik Lee , Haizhou Li

We present a novel model designed for resource-efficient multichannel speech enhancement in the time domain, with a focus on low latency, lightweight, and low computational requirements. The proposed model incorporates explicit spatial and…

声音 · 计算机科学 2024-01-17 Ashutosh Pandey , Buye Xu

The current dominant approach for neural speech enhancement is based on supervised learning by using simulated training data. The trained models, however, often exhibit limited generalizability to real-recorded data. To address this, this…

音频与语音处理 · 电气工程与系统科学 2025-03-25 Zhong-Qiu Wang

Far-field speech recognition in noisy and reverberant conditions remains a challenging problem despite recent deep learning breakthroughs. This problem is commonly addressed by acquiring a speech signal from multiple microphones and…

音频与语音处理 · 电气工程与系统科学 2018-10-17 Zhong Meng , Shinji Watanabe , John R. Hershey , Hakan Erdogan

We present a single-channel phase-sensitive speech enhancement algorithm that is based on modulation-domain Kalman filtering and on tracking the speech phase using circular statistics. With Kalman filtering, using that speech and noise are…

声音 · 计算机科学 2017-08-08 Nikolaos Dionelis , Mike Brookes

With the development of deep learning, speech enhancement has been greatly optimized in terms of speech quality. Previous methods typically focus on the discriminative supervised learning or generative modeling, which tends to introduce…

音频与语音处理 · 电气工程与系统科学 2025-10-31 Nan Xu , Zhaolong Huang , Xiaonan Zhi

Conventional far-field automatic speech recognition (ASR) systems typically employ microphone array techniques for speech enhancement in order to improve robustness against noise or reverberation. However, such speech enhancement techniques…

音频与语音处理 · 电气工程与系统科学 2021-12-23 Minhua Wu , Kenichi Kumatani , Shiva Sundaram , Nikko Strom , Bjorn Hoffmeister

Speech enhancement has benefited from the success of deep learning in terms of intelligibility and perceptual quality. Conventional time-frequency (TF) domain methods focus on predicting TF-masks or speech spectrum, via a naive convolution…

音频与语音处理 · 电气工程与系统科学 2020-09-24 Yanxin Hu , Yun Liu , Shubo Lv , Mengtao Xing , Shimin Zhang , Yihui Fu , Jian Wu , Bihong Zhang , Lei Xie

Many deep learning-based speech enhancement algorithms are designed to minimize the mean-square error (MSE) in some transform domain between a predicted and a target speech signal. However, optimizing for MSE does not necessarily guarantee…

声音 · 计算机科学 2020-01-31 Morten Kolbæk , Zheng-Hua Tan , Søren Holdt Jensen , Jesper Jensen

The most recent deep neural network (DNN) models exhibit impressive denoising performance in the time-frequency (T-F) magnitude domain. However, the phase is also a critical component of the speech signal that is easily overlooked. In this…

音频与语音处理 · 电气工程与系统科学 2021-06-10 Lu Zhang , Mingjiang Wang , Zehua Zhang , Xuyi Zhuang

Recently, our proposed recurrent neural network (RNN) based all deep learning minimum variance distortionless response (ADL-MVDR) beamformer method yielded superior performance over the conventional MVDR by replacing the matrix inversion…

声音 · 计算机科学 2021-04-27 Xiyun Li , Yong Xu , Meng Yu , Shi-Xiong Zhang , Jiaming Xu , Bo Xu , Dong Yu

Invariance to microphone array configuration is a rare attribute in neural beamformers. Filter-and-sum (FS) methods in this class define the target signal with respect to a reference channel. However, this not only complicates formulation…

音频与语音处理 · 电气工程与系统科学 2023-02-28 Anton Kovalyov , Kashyap Patel , Issa Panahi

Signal extraction from a single-channel mixture with additional undesired signals is most commonly performed using time-frequency (TF) masks. Typically, the mask is estimated with a deep neural network (DNN), and element-wise applied to the…

声音 · 计算机科学 2019-12-10 Wolfgang Mack , Emanuël A. P. Habets

Speaker verification aims to verify whether an input speech corresponds to the claimed speaker, and conventionally, this kind of system is deployed based on single-stream scenario, wherein the feature extractor operates in full frequency…

声音 · 计算机科学 2025-09-03 Wei Yao , Shen Chen , Jiamin Cui , Yaolin Lou