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Multi-frame approaches for single-microphone speech enhancement, e.g., the multi-frame minimum-power-distortionless-response (MFMPDR) filter, are able to exploit speech correlations across neighboring time frames. In contrast to…

Audio and Speech Processing · Electrical Eng. & Systems 2022-11-15 Marvin Tammen , Dörte Fischer , Bernd T. Meyer , Simon Doclo

This paper introduces a novel method to separate noisy speech into low or high frequency frames, in order to improve fundamental frequency (F0) estimation accuracy. In this proposal, the target signal is analyzed by means of the ensemble…

Audio and Speech Processing · Electrical Eng. & Systems 2021-12-21 A. Queiroz , R. Coelho

Linear prediction (LP) is an ubiquitous analysis method in speech processing. Various studies have focused on sparse LP algorithms by introducing sparsity constraints into the LP framework. Sparse LP has been shown to be effective in…

Audio and Speech Processing · Electrical Eng. & Systems 2020-06-09 Thomas Drugman

Speaker separation refers to isolating speech of interest in a multi-talker environment. Most methods apply real-valued Time-Frequency (T-F) masks to the mixture Short-Time Fourier Transform (STFT) to reconstruct the clean speech. Hence…

Audio and Speech Processing · Electrical Eng. & Systems 2020-04-16 Zhaoheng Ni , Michael I Mandel

Mainstream deep learning-based dysarthric speech detection approaches typically rely on processing the magnitude spectrum of the short-time Fourier transform of input signals, while ignoring the phase spectrum. Although considerable insight…

Audio and Speech Processing · Electrical Eng. & Systems 2022-01-25 Parvaneh Janbakhshi , Ina Kodrasi

Modern neural speech enhancement models usually include various forms of phase information in their training loss terms, either explicitly or implicitly. However, these loss terms are typically designed to reduce the distortion of phase…

Sound · Computer Science 2022-02-25 Doyeon Kim , Hyewon Han , Hyeon-Kyeong Shin , Soo-Whan Chung , Hong-Goo Kang

In this paper, we propose a neural-based coding scheme in which an artificial neural network is exploited to automatically compress and decompress speech signals by a trainable approach. Having a two-stage training phase, the system can be…

Sound · Computer Science 2016-01-25 Mahmood Yousefi-Azar , Farbod Razzazi

This paper proposes a dual-stage, low complexity, and reconfigurable technique to enhance the speech contaminated by various types of noise sources. Driven by input data and audio contents, the proposed dual-stage speech enhancement…

Audio and Speech Processing · Electrical Eng. & Systems 2021-05-18 Jun Yang , Nico Brailovsky

This study investigates phase reconstruction for deep learning based monaural talker-independent speaker separation in the short-time Fourier transform (STFT) domain. The key observation is that, for a mixture of two sources, with their…

Sound · Computer Science 2018-11-26 Zhong-Qiu Wang , Ke Tan , DeLiang Wang

Recently, deep neural networks (DNNs) have been successfully used for speech enhancement, and DNN-based speech enhancement is becoming an attractive research area. While time-frequency masking based on the short-time Fourier transform…

Audio and Speech Processing · Electrical Eng. & Systems 2020-08-21 Yuichiro Koyama , Tyler Vuong , Stefan Uhlich , Bhiksha Raj

In this work, we propose a novel consistency-preserving loss function for recovering the phase information in the context of phase reconstruction (PR) and speech enhancement (SE). Different from conventional techniques that directly…

Audio and Speech Processing · Electrical Eng. & Systems 2024-09-25 Pin-Jui Ku , Chun-Wei Ho , Hao Yen , Sabato Marco Siniscalchi , Chin-Hui Lee

This paper proposes an end-to-end approach for single-channel speaker-independent multi-speaker speech separation, where time-frequency (T-F) masking, the short-time Fourier transform (STFT), and its inverse are represented as layers within…

Sound · Computer Science 2018-04-30 Zhong-Qiu Wang , Jonathan Le Roux , DeLiang Wang , John R. Hershey

We address the problem of recovering a signal (up to global phase) from its short-time Fourier transform (STFT) magnitude measurements. This problem arises in several applications, including optical imaging and speech processing. In this…

Information Theory · Computer Science 2015-10-06 Tamir Bendory , Yonina C. Eldar

Traditional speech enhancement techniques modify the magnitude of a speech in time-frequency domain, and use the phase of a noisy speech to resynthesize a time domain speech. This work proposes a complex-valued Gaussian process latent…

Sound · Computer Science 2017-01-02 Sih-Huei Chen , Yuan-Shan Lee , Jia-Ching Wang

In our previous work, we proposed a neural vocoder called APNet, which directly predicts speech amplitude and phase spectra with a 5 ms frame shift in parallel from the input acoustic features, and then reconstructs the 16 kHz speech…

Audio and Speech Processing · Electrical Eng. & Systems 2023-11-21 Hui-Peng Du , Ye-Xin Lu , Yang Ai , Zhen-Hua Ling

This paper proposes a novel bidirectional neural vocoder, named BiVocoder, capable both of feature extraction and reverse waveform generation within the short-time Fourier transform (STFT) domain. For feature extraction, the BiVocoder takes…

Audio and Speech Processing · Electrical Eng. & Systems 2024-06-05 Hui-Peng Du , Ye-Xin Lu , Yang Ai , Zhen-Hua Ling

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

Recently, we proposed short-time Fourier transform (STFT)-based loss functions for training a neural speech waveform model. In this paper, we generalize the above framework and propose a training scheme for such models based on spectral…

Audio and Speech Processing · Electrical Eng. & Systems 2019-04-09 Shinji Takaki , Hirokazu Kameoka , Junichi Yamagishi

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…

Sound · Computer Science 2017-08-08 Nikolaos Dionelis , Mike Brookes

Large language models (LLMs) have significantly advanced audio processing through audio codecs that convert audio into discrete tokens, enabling the application of language modeling techniques to audio data. However, audio codecs often…

Audio and Speech Processing · Electrical Eng. & Systems 2024-09-19 Edresson Casanova , Ryan Langman , Paarth Neekhara , Shehzeen Hussain , Jason Li , Subhankar Ghosh , Ante Jukić , Sang-gil Lee