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ORCID29th EUSIPCO 2021: Dublin, Ireland
- 29th European Signal Processing Conference, EUSIPCO 2021, Dublin, Ireland, August 23-27, 2021. IEEE 2021, ISBN 978-9-0827-9706-0
- Carmen D'Andrea, Giovanni Interdonato
, Stefano Buzzi:
User-centric Handover in mmWave Cell-Free Massive MIMO with User Mobility. 1-5 - Tony Colin, Thomas Delamotte, Andreas Knopp:
Distortions Characterization for Dynamic Carrier Allocation in Ultra High-Throughput Satellites. 1-5 - Julien Lesouple, Barbara Pilastre, Yoann Altmann, Jean-Yves Tourneret:
Robust Hypersphere Fitting from Noisy Data Using an EM Algorithm. 1-5 - Joerg Schmalenstroeer, Jens Heitkaemper, Joerg Ullmann, Reinhold Haeb-Umbach:
Open Range Pitch Tracking for Carrier Frequency Difference Estimation from HF Transmitted Speech. 1-5 - Daniel Haider, Péter Balázs, Nicki Holighaus:
Phase-Based Signal Representations for Scattering. 6-10 - Midia Yousefi, Dimitra Emmanouilidou:
Audio-based Toxic Language Classification using Self-attentive Convolutional Neural Network. 11-15 - Zied Mnasri, Stefano Rovetta, Francesco Masulli:
A Novel Pitch Detection Algorithm Based On Instantaneous Frequency. 16-20 - Nirmesh J. Shah, M. Ali Basha Shaik, P. Periyasamy, Hemant A. Patil, Vikram Vij:
Exploiting Phase-based Features for Whisper vs. Speech Classification. 21-25 - Beáta Lorincz, Adriana Stan, Mircea Giurgiu:
Speaker verification-derived loss and data augmentation for DNN-based multispeaker speech synthesis. 26-30 - Ajinkya Kulkarni, Vincent Colotte, Denis Jouvet:
Improving transfer of expressivity for end-to-end multispeaker text-to-speech synthesis. 31-35 - Clément Le Moine, Nicolas Obin, Axel Roebel:
Towards end-to-end F0 voice conversion based on Dual-GAN with convolutional wavelet kernels. 36-40 - Kihyuk Jeong, Huu-Kim Nguyen, Hong-Goo Kang:
A Fast and Lightweight Text-To-Speech Model with Spectrum and Waveform Alignment Algorithms. 41-45 - Dan Oneata, Adriana Stan, Horia Cucu:
Speaker disentanglement in video-to-speech conversion. 46-50 - Taiyo Mineo, Hayaru Shouno:
Improving Convergence Rate of Sign Algorithm using Natural Gradient Method. 51-55 - Lam Pham, Chris Baume, Qiuqiang Kong, Tassadaq Hussain, Wenwu Wang, Mark D. Plumbley:
An Audio-Based Deep Learning Framework For BBC Television Programme Classification. 56-60 - Bi-Cheng Yan, Berlin Chen:
End-to-End Mispronunciation Detection and Diagnosis From Raw Waveforms. 61-65 - Hanxin Zhu, Chuang Shi, Yue Wang:
F0-estimation-based primary ambient extraction for stereo signals. 66-70 - Yuya Hosoda, Arata Kawamura, Youji Iiguni:
Phase Reconstruction for Artificial Bandwidth Extension toward Musical Instrument Sound Signal. 71-75 - Irene Martín-Morató, Annamaria Mesaros:
What is the ground truth? Reliability of multi-annotator data for audio tagging. 76-80 - Maximilian Kentgens, Shahd Al Hares, Peter Jax:
On the Upscaling of Higher-Order Ambisonics Signals for Sound Field Translation. 81-85 - Tobias Kabzinski, Jérôme Biot, Peter Jax:
A Model-Based 2-D Head-Tracking Method Using Microphones at the Ears. 86-90 - Kenta Imaizumi, Kenta Niwa, Kimitaka Tsutsumi:
Loudspeaker Array to Maximize Acoustic Contrast Using Proximal Splitting Method. 91-95 - Adrian Herzog, Emanuël A. P. Habets:
Signal-Dependent Mixing for Direction-Preserving Multichannel Noise Reduction. 96-100 - Robbe Van Rompaey, Marc Moonen:
Sound Zoning in an Ad-hoc Wireless Acoustic Sensor and Actuator Network. 101-105 - Christoph Hold, Archontis Politis, Leo McCormack, Ville Pulkki:
Spatial Filter Bank Design in the Spherical Harmonic Domain. 106-110 - Martin Jälmby, Filip Elvander, Toon van Waterschoot:
Low-Rank Tensor Modeling of Room Impulse Responses. 111-115 - Fabrice Katzberg, Marco Maaß, Alfred Mertins:
Coherence Based Trajectory Optimization for Compressive Sensing of Sound Fields. 116-120 - Marco Olivieri, Mirco Pezzoli, Fabio Antonacci, Augusto Sarti:
Near field Acoustic Holography on arbitrary shapes using Convolutional Neural Network. 121-125 - Kuldeep Khoria, Ankur T. Patil, Hemant A. Patil:
Significance of Constant-Q Transform for Voice Liveness Detection. 126-130 - Premjeet Singh, Goutam Saha, Md. Sahidullah:
Deep scattering network for speech emotion recognition. 131-135 - Shujie Zhao, Yan Yang, Israel Cohen, Lijun Zhang:
Speech Emotion Recognition Using Auditory Spectrogram and Cepstral Features. 136-140 - Mohammad MohammadAmini, Driss Matrouf, Jean-François Bonastre, Romain Serizel, Sandipana Dowerah, Denis Jouvet:
Compensate multiple distortions for speaker recognition systems. 141-145 - Shrishti Singh, Kuldeep Khoria, Hemant A. Patil:
Modified Group Delay Cepstral Coefficients for Voice Liveness Detection. 146-150 - Florian Hilgemann, Johannes Fabry, Peter Jax:
Design of IIR Filters for Active Noise Control by Constrained Optimization. 151-155 - Stefano Nobili, Valeria Bruschi, Ferruccio Bettarelli, Stefania Cecchi:
An Efficient Active Noise Control System with Online Secondary Path Estimation for Snoring Reduction. 156-160 - Marcel Nophut, Robert Hupke, Stephan Preihs, Jürgen Peissig:
Towards Wave-Domain Adaptive Filtering for Multichannel Acoustic Echo Cancellation in Higher-Order Ambisonics Systems. 161-165 - Paul M. Baggenstoss, Karl-Heinz Frommolt, Olaf Jahn, Frank Kurth:
Separation of Bird Calls and DOA estimation using a 4-Microphone Array. 166-170 - Clément Gaultier, Alexandre Guérin, Grégory Pallone, Marc Emerit:
Double-Talk Robust Acoustic Echo Cancellation Using Partition Block Frequency-Domain Adaptive Filtering. 171-175 - Abraham Woubie, Lauri Koivisto, Tom Bäckström:
Voice-quality Features for Deep Neural Network Based Speaker Verification Systems. 176-180 - Suradej Duangpummet, Jessada Karnjana, Waree Kongprawechnon, Masashi Unoki:
Blind Estimation of Room Acoustic Parameters and Speech Transmission Index using MTF-based CNNs. 181-185 - Emir Demirel, Sven Ahlbäck, Simon Dixon:
Computational Pronunciation Analysis in Sung Utterances. 186-190 - Venkata Srikanth Nallanthighal, Aki Härmä, Helmer Strik, Mathew Magimai-Doss:
Phoneme Based Respiratory Analysis of Read Speech. 191-195 - Ali Dehghan Firoozabadi, Pablo Irarrazaval, Pablo Adasme, David Zabala-Blanco, Pablo Palacios Játiva, Hugo Durney, Miguel Sanhueza-Olave, Cesar A. Azurdia-Meza:
Three-dimensional sound source localization by distributed microphone arrays. 196-200 - Arjun Venkat Venkatakrishnan, Pasi Pertilä, Mikko Parviainen:
Tampere University Rotated Circular Array Dataset. 201-205 - Leo McCormack, Archontis Politis, Simo Särkkä, Ville Pulkki:
Real-Time Tracking of Multiple Acoustical Sources Utilising Rao-Blackwellised Particle Filtering. 206-210 - Nils Poschadel, Robert Hupke, Stephan Preihs, Jürgen Peissig:
Direction of Arrival Estimation of Noisy Speech using Convolutional Recurrent Neural Networks with Higher-Order Ambisonics Signals. 211-215 - Ofer Schwartz:
Speaker Localization using Frobenius Norm with a Focus on Close Speaker and Noise Source. 216-220 - Elisa Tengan, Maja Taseska, Thomas Dietzen, Toon van Waterschoot:
Direction-of-arrival and power spectral density estimation using a single directional microphone. 221-225 - Graziano A. Manduzio, Nicola Forti, Roberto Sabatini, Paolo Braca, Giorgio Battistelli, Luigi Chisci:
Dynamic Source Localization via Finite-Element Underwater Acoustic Field Estimation. 226-230 - Pierre-Amaury Grumiaux, Srdan Kitic, Laurent Girin, Alexandre Guérin:
Improved feature extraction for CRNN-based multiple sound source localization. 231-235 - Daniel Krause, Archontis Politis, Konrad Kowalczyk:
Data Diversity for Improving DNN-based Localization of Concurrent Sound Events. 236-240 - Daniel Fejgin, Simon Doclo:
Comparison of Binaural RTF-Vector-Based Direction of Arrival Estimation Methods Exploiting an External Microphone. 241-245 - Gal Itzhak, Israel Cohen, Jacob Benesty:
Robust Differential Beamforming with Rectangular Arrays. 246-250 - Gal Itzhak, Jacob Benesty, Israel Cohen:
Quadratic Beamforming for Magnitude Estimation. 251-255 - Vasudha Sathyapriyan, Metin Çalis, Richard C. Hendriks:
Binaural beam-forming with dominant spatial cue preservation for hearing aids. 256-260 - Anna Barnov, Alex Gendelman, Amos Schreibman, Eli Tzirkel-Hancock, Sharon Gannot:
A Robust RLS Implementation of the ANC Block in GSC Structures. 261-265 - Tomohiro Nakatani, Rintaro Ikeshita, Naoyuki Kamo, Keisuke Kinoshita, Shoko Araki, Hiroshi Sawada:
Switching Convolutional Beamformer. 266-270 - Ali Dehghan Firoozabadi, Pablo Irarrazaval, Pablo Adasme, David Zabala-Blanco, Pablo Palacios Játiva, Hugo Durney, Miguel Sanhueza, Cesar A. Azurdia-Meza:
Speakers counting by proposed nested microphone array in combination with limited space SRP. 271-275 - Jacob Donley, Vladimir Tourbabin, Boaz Rafaely, Ravish Mehra:
Adaptive Multi-Channel Signal Enhancement Based on Multi-Source Contribution Estimation. 276-280 - Taiga Kawamura, Kohei Yatabe, Ryoichi Miyazaki:
Sparse Distortionless Beamformer Based on Nonconvex Optimization. 281-285 - Clément Dorffer, Thomas Paviet-Salomon, Gilles Le Chenadec, Angélique Drémeau:
Modal estimation in underwater acoustics by data-driven structured sparse decompositions. 286-290 - Yoshiaki Bando, Kouhei Sekiguchi, Kazuyoshi Yoshii:
Gamma Process FastMNMF for Separating an Unknown Number of Sound Sources. 291-295 - Lei Wang, Jie Zhu, Ina Kodrasi:
Multi-task Single Channel Speech Enhancement Using Speech Presence Probability As A Secondary Task Training Target. 296-300 - Shanshan Wang, Gaurav Naithani, Archontis Politis, Tuomas Virtanen:
Deep Neural Network Based Low-Latency Speech Separation with Asymmetric Analysis-Synthesis Window Pair. 301-305 - María Auxiliadora Sarmiento-Vega, Iván Durán-Díaz, Irene Fondón, Sergio Cruces:
Generalization of an Active Set Newton Algorithm with Alpha-Beta divergences for audio separation. 306-310 - Julian Neri, Roland Badeau, Philippe Depalle:
Unsupervised Blind Source Separation with Variational Auto-Encoders. 311-315 - Yoshiki Masuyama, Tomoro Tanaka, Kohei Yatabe, Tsubasa Kusano, Yasuhiro Oikawa:
Simultaneous Declipping and Beamforming via Alternating Direction Method of Multipliers. 316-320 - Koichi Saito, Tomohiko Nakamura, Kohei Yatabe, Yuma Koizumi, Hiroshi Saruwatari:
Sampling-Frequency-Independent Audio Source Separation Using Convolution Layer Based on Impulse Invariant Method. 321-325 - Naoki Narisawa, Rintaro Ikeshita, Norihiro Takamune, Daichi Kitamura, Tomohiko Nakamura, Hiroshi Saruwatari, Tomohiro Nakatani:
Independent Deeply Learned Tensor Analysis for Determined Audio Source Separation. 326-330 - Takuya Hasumi, Tomohiko Nakamura, Norihiro Takamune, Hiroshi Saruwatari, Daichi Kitamura, Yu Takahashi, Kazunobu Kondo:
Empirical Bayesian Independent Deeply Learned Matrix Analysis For Multichannel Audio Source Separation. 331-335 - Santiago Ruiz, Thomas Dietzen, Toon van Waterschoot, Marc Moonen:
A comparison between overlap-save and weighted overlap-add filter banks for multi-channel Wiener filter based noise reduction. 336-340 - Christos Garoufis, Athanasia Zlatintsi, Petros Maragos:
HTMD-Net: A Hybrid Masking-Denoising Approach to Time-Domain Monaural Singing Voice Separation. 341-345 - Alexander Bohlender, Ann Spriet, Wouter Tirry, Nilesh Madhu:
Neural Networks Using Full-Band and Subband Spatial Features for Mask Based Source Separation. 346-350 - Benjamin Stahl, Alois Sontacchi:
Speech Enhancement Quality Assessment Based on Aspect-Specific Qualities: A Preliminary Analysis. 351-355 - Florian Henkel, Gerhard Widmer:
Multi-modal Conditional Bounding Box Regression for Music Score Following. 356-360 - Jakob Abeßer, Jaydeep Chauhan, Prateek Pradeep Pillai, Michael Taenzer, Stylianos I. Mimilakis:
Predominant Jazz Instrument Recognition: Empirical Studies on Neural Network Architectures. 361-365 - Charles Brazier, Gerhard Widmer:
Handling Structural Mismatches in Real-time Opera Tracking. 366-370 - Johannes Gauer, Diana Kleingarn, Rainer Martin:
Analysis and Improvements of the Cepstrum Method for Fundamental Frequency Estimation in Music Signals. 371-375 - Andrew Wise, Anthony S. Maida, Ashok Kumar:
Attention Augmented CNNs for Musical Instrument Identification. 376-380 - Yudong Zhao, Changhong Wang, György Fazekas, Emmanouil Benetos, Mark B. Sandler:
Violinist identification based on vibrato features. 381-385 - Luís Carvalho, Gerhard Widmer:
Exploiting Temporal Dependencies for Cross-modal Music Piece Identification. 386-390 - Ching-Yu Chiu, Joann Ching, Wen-Yi Hsiao, Yu-Hua Chen, Alvin Wen-Yu Su, Yi-Hsuan Yang:
Source Separation-based Data Augmentation for Improved Joint Beat and Downbeat Tracking. 391-395 - Chih-Wei Wu, Phillip A. Williams, William Wolcott:
A Multitask Teacher-Student Framework for Perceptual Audio Quality Assessment. 396-400 - Amit Sofer, Tomás Kounovský, Jaroslav Cmejla, Zbynek Koldovský, Sharon Gannot:
Robust Relative Transfer Function Identification on Manifolds for Speech Enhancement. 401-405 - Pasi Pertilä, Emre Cakir, Aapo Hakala, Eemi Fagerlund, Tuomas Virtanen, Archontis Politis, Antti J. Eronen:
Mobile Microphone Array Speech Detection and Localization in Diverse Everyday Environments. 406-410 - Sherif Abdulatif, Karim Armanious, Jayasankar T. Sajeev, Karim Guirguis, Bin Yang:
Investigating Cross-Domain Losses for Speech Enhancement. 411-415 - Zuzana Jelcicová, Rasmus Jones, David Thorn Blix, Marian Verhelst, Jens Sparsø:
PeakRNN and StatsRNN: Dynamic Pruning in Recurrent Neural Networks. 416-420 - Sebastian Braun, Ivan Tashev:
On training targets for noise-robust voice activity detection. 421-425 - Shakeel A. Sheikh, Md. Sahidullah, Fabrice Hirsch, Slim Ouni:
StutterNet: Stuttering Detection Using Time Delay Neural Network. 426-430 - Aneesh Vartakavi, Amanmeet Garg, Zafar Rafii:
Audio Summarization for Podcasts. 431-435 - Takuya Fujimura, Yuma Koizumi, Kohei Yatabe, Ryoichi Miyazaki:
Noisy-target Training: A Training Strategy for DNN-based Speech Enhancement without Clean Speech. 436-440 - Lukás Samuel Marták, Rainer Kelz, Gerhard Widmer:
Probabilistic Modelling of Signal Mixtures with Differentiable Dictionaries. 441-445 - Pierre Beckmann, Mikolaj Kegler, Milos Cernak:
Word-Level Embeddings for Cross-Task Transfer Learning in Speech Processing. 446-450 - Lars Thieling, Peter Jax:
Generally Applicable Deep Speech Inpainting Using the Example of Bandwidth Extension. 451-455 - Tomoki Hayashi, Takenori Yoshimura, Masaya Inuzuka, Ibuki Kuroyanagi, Osamu Segawa:
Spontaneous Speech Summarization: Transformers All The Way Through. 456-460 - Marcin Witkowski, Magdalena Rybicka, Konrad Kowalczyk:
Sparse Linear Prediction-based Dereverberation for Signal Enhancement in Distant Speaker Verification. 461-465 - Andong Li, Chengshi Zheng, Lu Zhang, Xiaodong Li:
Learning to Inference with Early Exit in the Progressive Speech Enhancement. 466-470 - Mohamed Nabih Ali, Veronica Juliana Schmalz, Alessio Brutti, Daniele Falavigna:
A Speech Enhancement Front-End for Intent Classification in Noisy Environments. 471-475 - Qiongqiong Wang, Kong Aik Lee, Takafumi Koshinaka, Koji Okabe, Hitoshi Yamamoto:
Task-aware Warping Factors in Mask-based Speech Enhancement. 476-480 - Eike Jannik Nustede, Jörn Anemüller:
Towards speech enhancement using a variational U-Net architecture. 481-485 - Ju Lin, Adriaan J. de Lind van Wijngaarden, Melissa C. Smith, Kuang-Ching Wang:
Speaker-Aware Speech Enhancement with Self-Attention. 486-490 - Ryandhimas E. Zezario, Chiou-Shann Fuh, Hsin-Min Wang, Yu Tsao:
Speech Enhancement with Zero-Shot Model Selection. 491-495 - Yu-Wen Chen, Kuo-Hsuan Hung, Shang-Yi Chuang, Jonathan Sherman, Xugang Lu, Yu Tsao:
A Study of Incorporating Articulatory Movement Information in Speech Enhancement. 496-500 - Mickael Rouvier, Pierre-Michel Bousquet, Jarod Duret:
Study On the Temporal Pooling Used In Deep Neural Networks For Speaker Verification. 501-505 - Alexandros Koumparoulis, Gerasimos Potamianos, Samuel Thomas, Edmilson da Silva Morais:
Resource-efficient TDNN Architectures for Audio-visual Speech Recognition. 506-510 - Dipesh K. Singh, Preet P. Amin, Hardik B. Sailor, Hemant A. Patil:
Data Augmentation Using CycleGAN for End-to-End Children ASR. 511-515 - Tirthankar Banerjee, Narasimha Rao Thurlapati, V. Pavithra, S. Mahalakshmi, Dhanya Eledath, V. Ramasubramanian:
Few-Shot learning for frame-Wise phoneme recognition: Adaptation of matching networks. 516-520 - Zainab Alhakeem, Yoohwan Kwon, Hong-Goo Kang:
Disentangled Representations for Arabic Dialect Identification based on Supervised Clustering with Triplet Loss. 526-530 - Dhanya Eledath, P. Inbarajan, Anurag Biradar, Sathwick Mahadeva, V. Ramasubramanian:
End-to-end speech recognition from raw speech: Multi time-frequency resolution CNN architecture for efficient representation learning. 536-540 - Vincent P. Martin, Jean-Luc Rouas, Florian Boyer, Pierre Philip:
Automatic Speech Recognition systems errors for accident-prone sleepiness detection through voice. 541-545 - Spandan Dey, Goutam Saha, Md. Sahidullah:
Cross-Corpora Language Recognition: A Preliminary Investigation with Indian Languages. 546-550 - Jakob Abeßer, Saichand Gourishetti, András Kátai, Tobias Clauß, Prachi Sharma, Judith Liebetrau:
IDMT-Traffic: An Open Benchmark Dataset for Acoustic Traffic Monitoring Research. 551-555 - David S. Johnson, Wolfgang Lorenz, Michael Taenzer, Stylianos I. Mimilakis, Sascha Grollmisch, Jakob Abeßer, Hanna M. Lukashevich:
DESED-FL and URBAN-FL: Federated Learning Datasets for Sound Event Detection. 556-560 - Slobodan Djukanovic,