Results 21 to 30 of about 192 (149)

Head‐related transfer function–reserved time‐frequency masking for robust binaural sound source localization

open access: yesCAAI Transactions on Intelligence Technology, Volume 7, Issue 1, Page 26-33, March 2022., 2022
Abstract Various time‐frequency (T‐F) masks are being applied to sound source localization tasks. Moreover, deep learning has dramatically advanced T‐F mask estimation. However, existing masks are usually designed for speech separation tasks and are suitable only for single‐channel signals.
Hong Liu   +4 more
wiley   +1 more source

CycleGAN-based Unpaired Speech Dereverberation

open access: yesInterspeech 2022, 2022
Submitted to Interspeech ...
Hannah Muckenhirn   +6 more
openaire   +2 more sources

Speech dereverberation using a learned speech model [PDF]

open access: yes2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2015
We present a general single-channel speech dereverberation method based on an explicit generative model of reverberant and noisy speech. To regularize the model, we use a pre-learned speech model of clean and dry speech as a prior and perform posterior inference over the latent clean speech.
Dawen Liang   +2 more
openaire   +1 more source

Speech Dereverberation [PDF]

open access: yesThe Journal of the Acoustical Society of America, 1973
A method for dereverberating speech based on linear prediction is developed in detail. By properly analyzing and resynthesizing speech that has been reverberated by a real room, it is possible to remove most of the reverberation. The algorithm will be described and tapes will be played.
openaire   +1 more source

Speech Dereverberation in the STFT Domain

open access: yesCoRR, 2015
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 is constructed from a linear sum of nearby frames based on the channel impulse response.
Richard Stanton, Mike Brookes
openaire   +2 more sources

Speech intelligibility improvement in noisy reverberant environments based on speech enhancement and inverse filtering

open access: yesEURASIP Journal on Audio, Speech, and Music Processing, 2018
The speech intelligibility of indoor public address systems is degraded by reverberation and background noise. This paper proposes a preprocessing method that combines speech enhancement and inverse filtering to improve the speech intelligibility in such
Huan-Yu Dong, Chang-Myung Lee
doaj   +1 more source

Speech Dereverberation Based on Generative Adversarial Network with Additive Frequency Domain Decomposition

open access: yes工程科学与技术, 2022
The reverberant speech signal includes different frequency components induced by the effect of path delay. The frequency components are correlatedly modulated in frequency domain.
Haiyan QUAN, Tao WANG, Zhiqing ZHENG
doaj  

De-Noising Process in Room Impulse Response with Generalized Spectral Subtraction

open access: yesApplied Sciences, 2021
The generalized spectral subtraction algorithm (GBSS), which has extraordinary ability in background noise reduction, is historically one of the first approaches used for speech enhancement and dereverberation. However, the algorithm has not been applied
Min Chen, Chang-Myung Lee
doaj   +1 more source

Two-Microphone Dereverberation for Automatic Speech Recognition of Polish

open access: yesArchives of Acoustics, 2014
Reverberation is a common problem for many speech technologies, such as automatic speech recognition (ASR) systems. This paper investigates the novel combination of precedence, binaural and statistical independence cues for enhancing reverberant speech ...
Mikolaj KUNDEGORSKI   +2 more
doaj   +1 more source

Expectation‐maximisation for speech source separation using convolutive transfer function

open access: yesCAAI Transactions on Intelligence Technology, Volume 4, Issue 1, Page 47-53, March 2019., 2019
This study addresses the problem of under‐determined speech source separation from multichannel microphone signals, i.e. the convolutive mixtures of multiple sources. The time‐domain signals are first transformed to the short‐time Fourier transform (STFT) domain.
Xiaofei Li, Laurent Girin, Radu Horaud
wiley   +1 more source

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