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Open AccessJournal ArticleDOI

Time-Frequency Masking in the Complex Domain for Speech Dereverberation and Denoising

TLDR
This paper performs dereverberation and denoising using supervised learning with a deep neural network and defines the complex ideal ratio mask so that direct speech results after the mask is applied to reverberant and noisy speech.
Abstract
In real-world situations, speech is masked by both background noise and reverberation, which negatively affect perceptual quality and intelligibility. In this paper, we address monaural speech separation in reverberant and noisy environments. We perform dereverberation and denoising using supervised learning with a deep neural network. Specifically, we enhance the magnitude and phase by performing separation with an estimate of the complex ideal ratio mask. We define the complex ideal ratio mask so that direct speech results after the mask is applied to reverberant and noisy speech. Our approach is evaluated using simulated and real room impulse responses, and with background noises. The proposed approach improves objective speech quality and intelligibility significantly. Evaluations and comparisons show that it outperforms related methods in many reverberant and noisy environments.

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Citations
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Proceedings ArticleDOI

Data-driven Design of Perfect Reconstruction Filterbank for DNN-based Sound Source Enhancement

TL;DR: In this paper, the warped filterbank frame (WFBF) is considered as PRFB and the frequency characteristic of learned WFBF was in between STFT and the wavelet transform, and its effectiveness was confirmed by comparison with a standard STFT-based DNN whose input feature is compressed into the mel scale.
Journal ArticleDOI

A new Genetic Algorithm based fusion scheme in monaural CASA system to improve the performance of the speech

TL;DR: A new method is proposed in this research work to obtain a T–F binary mask from the segments of unvoiced speech and the performance of the proposed GA based fusion scheme is evaluated using measures such as quality and intelligibility.
Proceedings ArticleDOI

Environment-Dependent Attention-Driven Recurrent Convolutional Neural Network for Robust Speech Enhancement.

TL;DR: The proposed end-to-end environment-dependent attention-driven approach integrates an attention mechanism into bidirectional long short-term memory to acquire the weighted dynamic context between consecutive frames and outperformed existing methods on REVERB challenge.
Proceedings ArticleDOI

End-to-End Sound Source Enhancement Using Deep Neural Network in the Modified Discrete Cosine Transform Domain

TL;DR: An end-to-end deep neural network (DNN)-based source enhancement on the basis of a time-frequency (T-F) mask processing in the modified discrete cosine transform (MDCT)-domain is presented.
Proceedings ArticleDOI

A consolidated view of loss functions for supervised deep learning-based speech enhancement

TL;DR: In this article, the authors investigated a wide variety of loss spectral functions for a recurrent neural network architecture suitable to operate in online frame-by-frame processing and found that combining magnitude-only with phase-aware objectives always leads to improvements, even when the phase is not enhanced.
References
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Proceedings Article

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Adaptive Subgradient Methods for Online Learning and Stochastic Optimization

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Multitask Learning

TL;DR: Multi-task Learning (MTL) as mentioned in this paper is an approach to inductive transfer that improves generalization by using the domain information contained in the training signals of related tasks as an inductive bias.
Journal ArticleDOI

Image method for efficiently simulating small‐room acoustics

TL;DR: The theoretical and practical use of image techniques for simulating the impulse response between two points in a small rectangular room, when convolved with any desired input signal, simulates room reverberation of the input signal.
Journal ArticleDOI

Perceptual linear predictive (PLP) analysis of speech

TL;DR: A new technique for the analysis of speech, the perceptual linear predictive (PLP) technique, which uses three concepts from the psychophysics of hearing to derive an estimate of the auditory spectrum, and yields a low-dimensional representation of speech.
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