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EURASIP Journal on Audio, Speech, and Music Processing, Volume 2021
Volume 2021, Number 1, December 2021
- Jorge Llombart
, Dayana Ribas, Antonio Miguel, Luis Vicente
Progressive loss functions for speech enhancement with deep neural networks. 1 - Yuval Dorfan, Boaz Schwartz, Sharon Gannot:
Forward-backward recursive expectation-maximization for concurrent speaker tracking. 2 - Vincent Lostanlen, Christian El-Hajj, Mathias Rossignol, Grégoire Lafay, Joakim Andén, Mathieu Lagrange
Time-frequency scattering accurately models auditory similarities between instrumental playing techniques. 3 - Yuki Saishu, Amir Hossein Poorjam
A CNN-based approach to identification of degradations in speech signals. 1-10 - Chunyan Ji, Thosini Bamunu Mudiyanselage, Yutong Gao, Yi Pan
A review of infant cry analysis and classification. 1-17 - Rajat Hebbar
Deep multiple instance learning for foreground speech localization in ambient audio from wearable devices. 1-8 - Sören Schulze
Sparse pursuit and dictionary learning for blind source separation in polyphonic music recordings. 1-25 - Bracha Laufer-Goldshtein, Ronen Talmon, Sharon Gannot
Audio source separation by activity probability detection with maximum correlation and simplex geometry. 1-16 - Norihide Kitaoka
Dynamic out-of-vocabulary word registration to language model for speech recognition. 1-8 - Randall Ali
An integrated MVDR beamformer for speech enhancement using a local microphone array and external microphones. 10 - Kacper Radzikowski
Accent modification for speech recognition of non-native speakers using neural style transfer. 11 - Beatriz Martínez-González, José M. Pardo
Analysis of transition cost and model parameters in speaker diarization for meetings. 12 - Sushmita Thakallapalli
NMF-weighted SRP for multi-speaker direction of arrival estimation: robustness to spatial aliasing while exploiting sparsity in the atom-time domain. 13 - Zonglong Bai
Acoustic DOA estimation using space alternating sparse Bayesian learning. 14 - Randall Ali
Correction to: An integrated MVDR beamformer for speech enhancement using a local microphone array and external microphones. 15 - Hodaya Hammer, Shlomo E. Chazan, Jacob Goldberger, Sharon Gannot
Dynamically localizing multiple speakers based on the time-frequency domain. 16 - Yuxuan Ke, Andong Li, Chengshi Zheng, Renhua Peng, Xiaodong Li:
Low-complexity artificial noise suppression methods for deep learning-based speech enhancement algorithms. 17 - Duowei Tang
End-to-end speech emotion recognition using a novel context-stacking dilated convolution neural network. 18 - Sichen Liu
Frequency-dependent auto-pooling function for weakly supervised sound event detection. 19 - Miao Liu, Jing Wang
Neural network-based non-intrusive speech quality assessment using attention pooling function. 20 - Marco Gimm
Residual feedback suppression with extended model-based postfilters. 21 - Yong Lü, Han Lin
Feature compensation based on independent noise estimation for robust speech recognition. 22 - Maoshen Jia
Multi-source localization by using offset residual weight. 23 - Ziyi Xu
Components loss for neural networks in mask-based speech enhancement. 24 - Tobias Gburrek, Joerg Schmalenstroeer, Reinhold Haeb-Umbach:
Geometry calibration in wireless acoustic sensor networks utilizing DoA and distance information. 25 - Lujun Li
Adversarial joint training with self-attention mechanism for robust end-to-end speech recognition. 26 - Gui-Xin Shi, Wei-Qiang Zhang
Timestamp-aligning and keyword-biasing end-to-end ASR front-end for a KWS system. 27 - Alexandru-Lucian Georgescu
Performance vs. hardware requirements in state-of-the-art automatic speech recognition. 28 - Linhui Sun
Single-channel speech enhancement based on joint constrained dictionary learning. 29 - Mina Mounir
Musical note onset detection based on a spectral sparsity measure. 30 - Masoud Geravanchizadeh
Feature compensation based on the normalization of vocal tract length for the improvement of emotion-affected speech recognition. 31 - Sujan Kumar Roy
A noise PSD estimation algorithm using derivative-based high-pass filter in non-stationary noise conditions. 32 - Nili Cohen, Gershon Hazan, Boaz Schwartz, Sharon Gannot
An online algorithm for echo cancellation, dereverberation and noise reduction based on a Kalman-EM Method. 33 - Yanhua Long
Pronunciation augmentation for Mandarin-English code-switching speech recognition. 34 - Hongsheng Chen, Guoliang Chen, Kai Chen, Jing Lu:
Nonlinear residual echo suppression based on dual-stream DPRNN. 35 - David Ackermann, Fabian Brinkmann, Franz Zotter, Malte Kob, Stefan Weinzierl:
Comparative evaluation of interpolation methods for the directivity of musical instruments. 36 - Johannes M. Arend
Efficient binaural rendering of spherical microphone array data by linear filtering. 37 - You-Siang Chen, Zi-Jie Lin, Mingsian R. Bai
A multichannel learning-based approach for sound source separation in reverberant environments. 38 - Diego Di Carlo
dEchorate: a calibrated room impulse response dataset for echo-aware signal processing. 39 - Fangkun Liu, Hui Wang, Renhua Peng, Chengshi Zheng, Xiaodong Li:
U2-VC: one-shot voice conversion using two-level nested U-structure. 40 - Yuancheng Luo:
Spherical harmonic covariance and magnitude function encodings for beamformer design. 41 - Zolzaya Byambadorj
Text-to-speech system for low-resource language using cross-lingual transfer learning and data augmentation. 42 - Ofer Schwartz, Sharon Gannot
A recursive expectation-maximization algorithm for speaker tracking and separation. 43 - Yuki Takashima, Ryoichi Takashima
Unsupervised domain adaptation for lip reading based on cross-modal knowledge distillation. 44 - Jiacheng Yao, Jing Zhang
Anchor voiceprint recognition in live streaming via RawNet-SA and gated recurrent unit. 45
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