AASP-P17: Speech enhancement 2; Dereverberation and RIR estimation 2
Fri, 19 Apr, 08:20 - 10:20 (UTC +9)
Location: Poster Zone 5C
Session Type: Poster
Session Chair: Tomohiro Nakatani, NTT Corporation
Track: Audio and Acoustic Signal Processing
Click the to view the manuscript on IEEE Xplore Open Preview
 

AASP-P17.1: PHASE RECONSTRUCTION IN SINGLE CHANNEL SPEECH ENHANCEMENT BASED ON PHASE GRADIENTS AND ESTIMATED CLEAN-SPEECH AMPLITUDES
Yanjue Song, Nilesh Madhu, Ghent University - imec, Belgium
 

AASP-P17.2: MULTI-CMGAN+/+: LEVERAGING MULTI-OBJECTIVE SPEECH QUALITY METRIC PREDICTION FOR SPEECH ENHANCEMENT
George Close, William Ravenscroft, Thomas Hain, Stefan Goetze, University of Sheffield, United Kingdom of Great Britain and Northern Ireland
 

AASP-P17.3: A Deep Representation Learning-based Speech Enhancement Method Using Complex Convolution Recurrent Variational Autoencoder
Yang Xiang, Jingguang Tian, Xinhui Hu, Xinkang Xu, ZhaoHui Yin, Hithink RoyalFlush AI Research Institute, China
 

AASP-P17.4: REMIXED2REMIXED: DOMAIN ADAPTATION FOR SPEECH ENHANCEMENT BY NOISE2NOISE LEARNING WITH REMIXING
Li Li, Shogo Seki, CyberAgent, Inc., Japan
 

AASP-P17.5: AN EMPIRICAL STUDY ON THE IMPACT OF POSITIONAL ENCODING IN TRANSFORMER-BASED MONAURAL SPEECH ENHANCEMENT
Qiquan Zhang, University of New South Wales, Australia; Meng Ge, Hongxu Zhu, National University of Singapore, Singapore; Eliathamby Ambikairajah, University of New South Wales, Australia; Qi Song, Alibaba, China; Zhaoheng Ni, Meta, United States of America; Haizhou Li, The Chinese University of Hong Kong, Shenzhen, China
 

AASP-P17.6: Spiking Structured State Space Model for Monaural Speech Enhancement
Yu Du, Tsinghua University, China; Xu Liu, Yansong CHUA, China Nanhu Academy of Electronics and Information Technology (CNAEIT), China

AASP-P17.7: PARAMETER ESTIMATION PROCEDURES FOR DEEP MULTI-FRAME MVDR FILTERING FOR SINGLE-MICROPHONE SPEECH ENHANCEMENT
Marvin Tammen, Simon Doclo, University of Oldenburg, Germany

AASP-P17.8: Masked spectrogram prediction for unsupervised domain adaptation in speech enhancement
Katerina Zmolikova, Michael Syskind Pedersen, Jesper Jensen, Demant A/S Ringgold standard institution Smorum Denmark andAalborg University Ringgold standard institution - Electronic Systems Aalborg Denmark
 

AASP-P17.9: MICROPHONE SUBSET SELECTION FOR THE WEIGHTED PREDICTION ERROR ALGORITHM USING A GROUP SPARSITY PENALTY
Anselm Lohmann, Carl von Ossietzky Universität Oldenburg, Germany; Toon van Waterschoot, KU Leuven, Belgium; Joerg Bitzer, Fraunhofer IDMT, Germany; Simon Doclo, Carl von Ossietzky Universität Oldenburg, Fraunhofer IDMT, Germany
 

AASP-P17.10: DUAL-PATH MINIMUM-PHASE AND ALL-PASS DECOMPOSITION NETWORK FOR SINGLE CHANNEL SPEECH DEREVERBERATION
Xi Liu, Szu-Jui Chen, John Hansen, The University of Texas at Dallas, United States of America

AASP-P17.11: Speech Dereverberation With Frequency Domain Autoregressive Modeling
Anurenjan Purushothaman, Government Engineering College Idukki, India; Debottam Dutta, University of Illinois Urbana-Champaign, India; Rohit Kumar, Johns Hopkins University, India; Sriram Ganapathy, IISc Bangalore, India

AASP-P17.12: A Flexible Framework for Expectation Maximization-Based MIMO System Identification for Time-Variant Linear Acoustic Systems
Tobias Kabzinski, Peter Jax, RWTH Aachen University Ringgold standard institution - Institute of Communication Systems Muffeter Weg 3a , Aachen 52056 Germany

AASP-P17.13: JOINT DEREVERBERATION AND BEAMFORMING WITH BLIND ESTIMATION OF THE SHAPE PARAMETER OF THE DESIRED SOURCE PRIOR
Shekhar Kumar Yadav, Nithin V. George, Indian Institute of Technology Gandhinagar, India