Jeih-Weih Hung

Orcid: 0000-0001-9366-3070

According to our database1, Jeih-Weih Hung authored at least 109 papers between 1998 and 2024.

Collaborative distances:
  • Dijkstra number2 of four.
  • Erdős number3 of four.

Timeline

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Bibliography

2024
What do neural networks listen to? Exploring the crucial bands in Speech Enhancement using Sinc-convolution.
CoRR, 2024

2023
Improving Speech Enhancement Performance by Leveraging Contextual Broad Phonetic Class Information.
IEEE ACM Trans. Audio Speech Lang. Process., 2023

NAaLOSS: Rethinking the Objective of Speech Enhancement.
Proceedings of the 33rd IEEE International Workshop on Machine Learning for Signal Processing, 2023

ConSep: a Noise- and Reverberation-Robust Speech Separation Framework by Magnitude Conditioning.
Proceedings of the 24th International Conference on Digital Signal Processing, 2023

Exploiting Discrete Wavelet Transform Features in Speech Enhancement Technique Adaptive FullSubNet+.
Proceedings of the International Conference on Consumer Electronics - Taiwan, 2023

Improving the performance of CMGAN in speech enhancement with the phone fortified perceptual loss.
Proceedings of the International Conference on Consumer Electronics - Taiwan, 2023

2022
Time-Reversal Enhancement Network With Cross-Domain Information for Noise-Robust Speech Recognition.
IEEE Multim., 2022

Bi-Sep: A Multi-Resolution Cross-Domain Monaural Speech Separation Framework.
Proceedings of the International Conference on Technologies and Applications of Artificial Intelligence, 2022

Adaptive-FSN: Integrating Full-Band Extraction and Adaptive Sub-Band Encoding for Monaural Speech Enhancement.
Proceedings of the IEEE Spoken Language Technology Workshop, 2022

Exploiting the compressed spectral loss for the learning of the DEMUCS speech enhancement network.
Proceedings of the 34th Conference on Computational Linguistics and Speech Processing, 2022

A Preliminary Study of the Application of Discrete Wavelet Transform Features in Conv-TasNet Speech Enhancement Model.
Proceedings of the 34th Conference on Computational Linguistics and Speech Processing, 2022

A Preliminary Study of Employing Lowpass-Filtered and Time-Reversed Feature Sequences as Data Augmentation for Speech Enhancement Deep Networks.
Proceedings of the International Symposium on Intelligent Signal Processing and Communication Systems, 2022

Improving the performance of DEMUCS in speech enhancement with the perceptual metric loss.
Proceedings of the IEEE International Conference on Consumer Electronics - Taiwan, 2022

2021
Employing low-pass filtered temporal speech features for the training of ideal ratio mask in speech enhancement.
Proceedings of the 33rd Conference on Computational Linguistics and Speech Processing, 2021

Cross-Domain Single-Channel Speech Enhancement Model with BI-Projection Fusion Module for Noise-Robust ASR.
Proceedings of the 2021 IEEE International Conference on Multimedia and Expo, 2021

The effect of reducing the acoustic-frequency resolution for spectrograms used in deep denoising auto-encoder.
Proceedings of the IEEE International Conference on Consumer Electronics-Taiwan, 2021

TENET: A Time-Reversal Enhancement Network for Noise-Robust ASR.
Proceedings of the IEEE Automatic Speech Recognition and Understanding Workshop, 2021

2020
Time-Domain Multi-Modal Bone/Air Conducted Speech Enhancement.
IEEE Signal Process. Lett., 2020

Speech enhancement guided by contextual articulatory information.
CoRR, 2020

Multi-view Attention-based Speech Enhancement Model for Noise-robust Automatic Speech Recognition.
Proceedings of the 32nd Conference on Computational Linguistics and Speech Processing, 2020

The preliminary study of robust speech feature extraction based on maximizing the accuracy of states in deep acoustic models.
Proceedings of the 32nd Conference on Computational Linguistics and Speech Processing, 2020

Incorporating Broad Phonetic Information for Speech Enhancement.
Proceedings of the Interspeech 2020, 2020

Exponentiated magnitude spectrogram-based relative-to-maximum masking for speech enhancement in adverse environments.
Proceedings of the IEEE International Conference on Consumer Electronics - Taiwan, 2020

Lowpass-filtered relative-to-maximum masking for speech enhancement in noise-corrupted environments.
Proceedings of the IEEE International Conference on Consumer Electronics - Taiwan, 2020

2019
Time-Domain Multi-modal Bone/air Conducted Speech Enhancement.
CoRR, 2019

Distributed Microphone Speech Enhancement based on Deep Learning.
CoRR, 2019

Speech enhancement based on the integration of fully convolutional network, temporal lowpass filtering and spectrogram masking.
Proceedings of the 31st Conference on Computational Linguistics and Speech Processing, 2019

Speaker-Aware Deep Denoising Autoencoder with Embedded Speaker Identity for Speech Enhancement.
Proceedings of the Interspeech 2019, 2019

An evaluation study of modulation-domain wavelet denoising method by alleviating different sub-band portions for speech enhancement.
Proceedings of the IEEE International Conference on Consumer Electronics - Taiwan, 2019

Smoothing the acoustic spectral time series of speech signals for noise reduction.
Proceedings of the IEEE International Conference on Consumer Electronics - Taiwan, 2019

2018
Suppression by Selecting Wavelets for Feature Compression in Distributed Speech Recognition.
IEEE ACM Trans. Audio Speech Lang. Process., 2018

Speech Enhancement Based on Reducing the Detail Portion of Speech Spectrograms in Modulation Domain via Discrete Wavelet Transform.
CoRR, 2018

Speech Enhancement Based on Reducing the Detail Portion of Speech Spectrograms in Modulation Domain via DiscreteWavelet Transform.
Proceedings of the 11th International Symposium on Chinese Spoken Language Processing, 2018

2017
多樣訊雜比之訓練語料於降噪自動編碼器其語音強化功能之初步研究 (A Preliminary Study of Various SNR-level Training Data in the Denoising Auto-encoder (DAE) Technique for Speech Enhancement) [In Chinese].
Proceedings of the 29th Conference on Computational Linguistics and Speech Processing, 2017

Wavelet Speech Enhancement Based on Robust Principal Component Analysis.
Proceedings of the Interspeech 2017, 2017

2016
Robust Speech Recognition via Enhancing the Complex-Valued Acoustic Spectrum in Modulation Domain.
IEEE ACM Trans. Audio Speech Lang. Process., 2016

Wavelet Speech Enhancement Based on Nonnegative Matrix Factorization.
IEEE Signal Process. Lett., 2016

Employing median filtering to enhance the complex-valued acoustic spectrograms in modulation domain for noise-robust speech recognition.
Proceedings of the 10th International Symposium on Chinese Spoken Language Processing, 2016

Speech enhancement via ensemble modeling NMF adaptation.
Proceedings of the IEEE International Conference on Consumer Electronics-Taiwan, 2016

Leveraging nonnegative matrix factorization in processing the temporal modulation spectrum for speech enhancement.
Proceedings of the IEEE International Conference on Consumer Electronics-Taiwan, 2016

Linear prediction filtering on cepstral time series for noise-robust speech recognition.
Proceedings of the IEEE International Conference on Consumer Electronics-Taiwan, 2016

2015
Histogram equalization of contextual statistics of speech features for robust speech recognition.
Multim. Tools Appl., 2015

Magnitude replacement of real and imaginary modulation spectrum of acoustic spectrograms for noise-robust speech recognition.
Proceedings of the IEEE International Conference on Consumer Electronics - Taiwan, 2015

Enhancing the complex-valued acoustic spectrograms in modulation domain for creating noise-robust features in speech recognition.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

2014
Sliding backstepping control design for robotic manipulator systems with motor dynamics.
Proceedings of the 11th IEEE International Conference on Control & Automation, 2014

Leveraging threshold denoising on DCT-based modulation spectrum for noise robust speech recognition.
Proceedings of the 11th IEEE International Conference on Control & Automation, 2014

Speech enhancement using segmental nonnegative matrix factorization.
Proceedings of the IEEE International Conference on Acoustics, 2014

Spatial histogram equalization of complex-valued acoustic spectra in modulation domain for noise-robust speech recognition.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2014

2013
Employing Linear Prediction Coding in Feature Time Sequences for Robust Speech Recognition in Noisy Environments.
Int. J. Comput. Linguistics Chin. Lang. Process., 2013

Improved DFT-Based Channel Estimation for TDS-OFDM Wireless Communication Systems.
IEICE Trans. Commun., 2013

Intra-frame cepstral sub-band weighting and histogram equalization for noise-robust speech recognition.
EURASIP J. Audio Speech Music. Process., 2013

雜訊環境下應用線性估測編碼於特徵時序列之強健性語音辨識 (Employing linear prediction coding in feature time sequences for robust speech recognition in noisy environments) [In Chinese].
Proceedings of the 25th Conference on Computational Linguistics and Speech Processing, 2013

分頻式調變頻譜分解於強健性語音辨識 (Sub-band modulation spectrum factorization in robust speech recognition) [In Chinese].
Proceedings of the 25th Conference on Computational Linguistics and Speech Processing, 2013

Histogram equalization of real and imaginary modulation spectra for noise-robust speech recognition.
Proceedings of the INTERSPEECH 2013, 2013

Filtering on the temporal probability sequence in histogram equalization for robust speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2013

Overlapped sub-band modulation spectrum normalization techniques for robust speech recognition.
Proceedings of the 10th International Conference on Fuzzy Systems and Knowledge Discovery, 2013

Robustifying cepstral features by mitigating the outlier effect for noisy speech recognition.
Proceedings of the 10th International Conference on Fuzzy Systems and Knowledge Discovery, 2013

Modulation spectrum power-law expansion for robust speech recognition.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2013

2012
Improved modulation spectrum enhancement methods for robust speech recognition.
Signal Process., 2012

Speech Recognition Leveraging Histogram Equalization Methods.
Int. J. Comput. Linguistics Chin. Lang. Process., 2012

Enhancing the Magnitude Spectrum of Speech Features for Robust Speech Recognition.
EURASIP J. Adv. Signal Process., 2012

改良式統計圖等化法強鍵性語音辨識之研究 (Improved Histogram Equalization Methods for Robust Speech Recognition) [In Chinese].
Proceedings of the 24th Conference on Computational Linguistics and Speech Processing, 2012

Modulation spectrum exponential weighting for robust speech recognition.
Proceedings of the 12th International Conference on ITS Telecommunications, 2012

A study on cepstral sub-band normalization for robust ASR.
Proceedings of the 8th International Symposium on Chinese Spoken Language Processing, 2012

Exploring Joint Equalization of Spatial-Temporal Contextual Statistics of Speech Features for Robust Speech Recognition.
Proceedings of the INTERSPEECH 2012, 2012

Leveraging gain normalization for sub-band temporal features in noise-robust speech recognition.
Proceedings of the 9th International Conference on Fuzzy Systems and Knowledge Discovery, 2012

2011
Compensating the Speech Features via Discrete Cosine Transform for Robust Speech Recognition (基於離散餘弦轉換之語音特徵的強健性補償法).
Proceedings of the 23rd Conference on Computational Linguistics and Speech Processing, 2011

機率式調變頻譜分解於強健性語音辨識 (Probabilistic Modulation Spectrum Factorization for Robust Speech Recognition) [In Chinese].
Proceedings of the Poster Proceedings of the 23rd Conference on Computational Linguistics and Speech Processing, 2011

Exploiting principal component analysis in modulation spectrum enhancement for robust speech recognition.
Proceedings of the Eighth International Conference on Fuzzy Systems and Knowledge Discovery, 2011

2010
進階式調變頻譜補償法於強健性語音辨識之研究 (Advanced Modulation Spectrum Compensation Techniques for Robust Speech Recognition) [In Chinese].
Proceedings of the 22th Conference on Computational Linguistics and Speech Processing, 2010

最小變異數調變頻譜濾波器於強健性語音辨識之研究 (A Study of Minimum Variance Modulation Filter for Robust Speech Recognition) [In Chinese].
Proceedings of the 22th Conference on Computational Linguistics and Speech Processing, 2010

DCT-based processing of dynamic features for robust speech recognition.
Proceedings of the 7th International Symposium on Chinese Spoken Language Processing, 2010

Magnitude spectrum enhancement for robust speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2010

2009
Incorporating Codebook and Utterance Information in Cepstral Statistics Normalization Techniques for Robust Speech Recognition in Additive Noise Environments.
IEEE Signal Process. Lett., 2009

Subband Feature Statistics Normalization Techniques Based on a Discrete Wavelet Transform for Robust Speech Recognition.
IEEE Signal Process. Lett., 2009

Study of Associative Cepstral Statistics Normalization Techniques for Robust Speech Recognition in Additive Noise Environments.
Int. J. Comput. Linguistics Chin. Lang. Process., 2009

強健性語音辨識中分頻段調變頻譜補償之研究 (A Study of Sub-band Modulation Spectrum Compensation for Robust Speech Recognition) [In Chinese].
Proceedings of the 21st Conference on Computational Linguistics and Speech Processing, 2009

併合式倒頻譜統計正規化技術於強健性語音辨識之研究 (A Study of Hybrid-based Cepstral Statistics Normalization Techniques for Robust Speech Recognition) [In Chinese].
Proceedings of the 21st Conference on Computational Linguistics and Speech Processing, 2009

強健性語音辨識中基於小波轉換之分頻統計補償技術的研究 (A Study of Sub-band Feature Statistics Compensation Techniques Based on a Discrete Wavelet Transform for Robust Speech Recognition) [In Chinese].
Proceedings of the 21st Conference on Computational Linguistics and Speech Processing, 2009

Integrating codebook and utterance information in cepstral statistics normalization techniques for robust speech recognition.
Proceedings of the INTERSPEECH 2009, 2009

Sub-band feature statistics compensation techniques based on discrete wavelet transform for robust speech recognition.
Proceedings of the 2009 IEEE International Conference on Multimedia and Expo, 2009

Sub-band modulation spectrum compensation for robust speech recognition.
Proceedings of the 2009 IEEE Workshop on Automatic Speech Recognition & Understanding, 2009

2008
Constructing Modulation Frequency Domain-Based Features for Robust Speech Recognition.
IEEE Trans. Speech Audio Process., 2008

Cepstral Statistics Compensation and Normalization Using Online Pseudo Stereo Codebooks for Robust Speech Recognition in Additive Noise Environments.
IEICE Trans. Inf. Syst., 2008

調變頻譜正規化法使用於強健語音辨識之研究 (Study of Modulation Spectrum Normalization Techniques for Robust Speech Recognition) [In Chinese].
Proceedings of the 20th Conference on Computational Linguistics and Speech Processing, 2008

組合式倒頻譜統計正規化法於強健性語音辨識之研究 (Associative Cepstral Statistics Normalization Techniques for Robust Speech Recognition) [In Chinese].
Proceedings of the 20th Conference on Computational Linguistics and Speech Processing, 2008

強健性語音辨識中能量相關特徵之改良式正規化技術的研究 (Study of the Improved Normalization Techniques of Energy-Related Features for Robust Speech Recognition) [In Chinese].
Proceedings of the 20th Conference on Computational Linguistics and Speech Processing, 2008

Silence feature normalization for robust speech recognition in additive noise environments.
Proceedings of the INTERSPEECH 2008, 2008

Improved modulation spectrum normalization techniques for robust speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2008

2007
端點偵測技術在強健語音參數擷取之研究 (Study of the Voice Activity Detection Techniques for Robust Speech Feature Extraction) [In Chinese].
Proceedings of the 19th Conference on Computational Linguistics and Speech Processing, 2007

加成性雜訊環境下運用特徵參數統計補償法於強健性語音辨識 (Feature Statistics Compensation for Robust Speech Recognition in Additive Noise Environments) [In Chinese].
Proceedings of the 19th Conference on Computational Linguistics and Speech Processing, 2007

Optimization of temporal filters in the modulation frequency domain for constructing robust features in speech recognition.
Proceedings of the INTERSPEECH 2007, 2007

Speech feature compensation based on pseudo stereo codebooks for robust speech recognition in additive noise environments.
Proceedings of the INTERSPEECH 2007, 2007

Optimization of Temporal Filters in the Modulation Frequency Domain via Constrained Linear Discriminant Analysis (C-LDA) for Constructing Robust Features in Speech Recognition.
Proceedings of the IEEE International Conference on Acoustics, 2007

2006
Optimization of temporal filters for constructing robust features in speech recognition.
IEEE Trans. Speech Audio Process., 2006

Silence energy normalization for robust speech recognition in additive noise environment.
Proceedings of the INTERSPEECH 2006, 2006

Cepstral Statistics Compensation Using Online Pseudo Stereo Codebooks for Robust Speech Recognition in Additive Noise Environments.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

2004
Data-driven temporal filters based on maximum mutual information for robust features in speech recognition.
Proceedings of the 2004 International Symposium on Chinese Spoken Language Processing, 2004

2003
Data-driven temporal filters based on multi-eigenvectors for robust features in speech recognition.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

2002
Data-driven temporal filters obtained via different optimization criteria evaluated on Aurora2 database.
Proceedings of the 7th International Conference on Spoken Language Processing, ICSLP2002, 2002

Data-driven temporal filters for robust features in speech recognition obtained via Minimum Classification Error (MCE).
Proceedings of the IEEE International Conference on Acoustics, 2002

2001
New approaches for domain transformation and parameter combination for improved accuracy in parallel model combination (PMC) techniques.
IEEE Trans. Speech Audio Process., 2001

Comparative analysis for data-driven temporal filters obtained via principal component analysis (PCA) and linear discriminant analysis (LDA) in speech recognition.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

2000
Automatic metric-based speech segmentation for broadcast news via principal component analysis.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

1999
Improved parallel model combination techniques with split Gaussian mixtures for speech recognition under noisy conditions.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

1998
Robust entropy-based endpoint detection for speech recognition in noisy environments.
Proceedings of the 5th International Conference on Spoken Language Processing, Incorporating The 7th Australian International Speech Science and Technology Conference, Sydney Convention Centre, Sydney, Australia, 30th November, 1998

Improved robust speech recognition considering signal correlation approximated by taylor series.
Proceedings of the 5th International Conference on Spoken Language Processing, Incorporating The 7th Australian International Speech Science and Technology Conference, Sydney Convention Centre, Sydney, Australia, 30th November, 1998

Improved parallel model combination based on better domain transformation for speech recognition under noisy environments.
Proceedings of the 5th International Conference on Spoken Language Processing, Incorporating The 7th Australian International Speech Science and Technology Conference, Sydney Convention Centre, Sydney, Australia, 30th November, 1998

Improved robustness for speech recognition under noisy conditions using correlated parallel model combination.
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998


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