Abstract is missing.
- Structured total least squares based internal delay estimation for distributed microphone auto-localizationJie Zhang, Richard C. Hendriks, Richard Heusdens. 1-5 [doi]
- Constrained multi-channel linear prediction for adaptive speech dereverberationAnte Jukic, Zichao Wang, Toon van Waterschoot, Timo Gerkmann, Simon Doclo. 1-5 [doi]
- Approximation of a nonlinear distortion function for combined linear and nonlinear residual echo suppressionIngo Schalk-Schupp, Friedrich Faubel, Markus Buck, Andreas Wendemuth. 1-5 [doi]
- Affine projection algorithm for acoustic feedback cancellation using prediction error method in hearing aidsLinh T. T. Tran, Hai Huyen Dam, Sven E. Nordholm. 1-5 [doi]
- Multichannel time delay estimation for acoustic source localization via robust adaptive blind system identificationHongsen He, Jingdong Chen, Jacob Benesty, Tao Yang. 1-5 [doi]
- Convolutive blind source separation with low latencyJiawen Chua, Ganlong Wang, W. Bastiaan Kleijn. 1-5 [doi]
- Discriminative and reconstructive basis training for audio source separation with semi-supervised nonnegative matrix factorizationDaichi Kitamura, Nobutaka Ono, Hiroshi Saruwatari, Yu Takahashi, Kazunobu Kondo. 1-5 [doi]
- Evaluation of spatial active noise cancellation performance using spherical harmonic analysisHanchi Chen, Jihui Zhang, Prasanga N. Samarasinghe, Thushara D. Abhayapala. 1-5 [doi]
- Dynamic group sparsity for non-negative matrix factorization with application to unsupervised source separationXu Li, Xiaofei Wang, Qiang Fu, Yonghong Yan 0002. 1-5 [doi]
- An informed separation algorithm based on sound field mapping for speech recognition systemsDejan Markovic, Jigyasa Popat, Fabio Antonacci, Augusto Sarti, T. Kishore Kumar. 1-5 [doi]
- Spherical harmonic rake receivers for dereverberationHamza A. Javed, Alastair H. Moore, Patrick A. Naylor. 1-5 [doi]
- Speech enhancement using arch modelAviva Atkins, Israel Cohen. 1-5 [doi]
- On the evaluation of multichannel blind system identification from the viewpoint of system equalizationWancheng Zhang, Patrick A. Naylor, Zunwen He, Yan Zhang. 1-4 [doi]
- Late reverberation PSD estimation for single-channel dereverberation using relative convolutive transfer functionsSebastian Braun, Boaz Schwartz, Sharon Gannot, Emanuel A. P. Habets. 1-5 [doi]
- Insight into linear periodically time-varying coherence reduction methods for stereophonic acoustic echo cancellationMaria Luis Valero, Emanuel A. P. Habets. 1-5 [doi]
- Direction-of-arrival estimation based on joint diagonalization of matrices in different direct-to-reverberation ratiosRyusuke Tanaka, Yoichi Haneda. 1-5 [doi]
- A multiframe parametric wiener filter for acoustic echo suppressionHai Huang, Christian Hofmann, Walter Kellermann, Jingdong Chen, Jacob Benesty. 1-5 [doi]
- Performance analysis of a dual microphone superdirective beamformer and approximate expressions for the near-field propagation regimeShmulik Markovich Golan, Dovid Y. Levin, Sharon Gannot. 1-5 [doi]
- A contingency multi-microphone noise reduction strategy based on linearly constrained Multi-channel Wiener filteringRandall Ali, Marc Moonen. 1-4 [doi]
- Robust superdirective beamformer with optimal regularizationAviva Atkins, Yuval Ben-Hur, Israel Cohen, Jacob Benesty. 1-5 [doi]
- Artificial bandwidth extension using deep neural networks for spectral envelope estimationJohannes Abel, Maximilian Strake, Tim Fingscheidt. 1-5 [doi]
- An experimental study of noise on the performance of a low bit rate parametric speech coderWenhua Shi, Xiongwei Zhang, Xia Zou, Wei Han. 1-4 [doi]
- Student's t multichannel nonnegative matrix factorization for blind source separationKoichi Kitamura, Yoshiaki Bando, Katsutoshi Itoyama, Kazuyoshi Yoshii. 1-5 [doi]
- Linear prediction based dereverberation for spherical microphone arraysAlastair H. Moore, Patrick A. Naylor. 1-5 [doi]
- The block-sparse proportionate second-order volterra filtering algorithms for nonlinear echo cancellationJianming Liu, Quintin Liu, Steven L. Grant, Yahong Rosa Zheng. 1-5 [doi]
- Approximate MVDR and MMSE beamformers exploiting scale-invariant reconstruction of signals on microphonesZbynek Koldovský, Francesco Nesta. 1-5 [doi]
- Room transfer function measurement from a directional loudspeakerPrasanga N. Samarasinghe, Thushara D. Abhayapala. 1-5 [doi]
- PSD estimation in beamspace using property of M-matrixKenta Niwa, Tomoko Kawase, Kazunori Kobayashi, Yusuke Hioka. 1-5 [doi]
- Immersive audio for human-machine interface of unmanned ground vehiclesVinay K. Kothapally, Steven L. Grant, Yahong Rosa Zheng. 1-5 [doi]
- On directivity of a circular array with directional microphonesHongsen He, Xiaojun Qiu, Tao Yang. 1-5 [doi]
- Modeling audio directional statistics using a probabilistic spatial dictionary for speaker diarization in real meetingsMahmoud Fakhry, Nobutaka Ito, Shoko Araki, Tomohiro Nakatani. 1-5 [doi]
- Ego-noise reduction for a hose-shaped rescue robot using determined rank-1 multichannel nonnegative matrix factorizationMoe Takakusaki, Daichi Kitamura, Nobutaka Ono, Takeshi Yamada, Shoji Makino, Hiroshi Saruwatari. 1-4 [doi]
- Deep sparse rectifier neural networks for speech denoisingLie Xu, Chiu-sing Choy, Yi-Wen Li. 1-5 [doi]
- New method for synthesizing personalized head-related transfer functionLei Wang, Xiangyang Zeng. 1-5 [doi]
- An improved soft decision based noise power estimation employing adaptive prior and conditional smoothingPei Chee Yong, Sven Nordholm. 1-5 [doi]
- A phoneme-based pre-training approach for deep neural network with application to speech enhancementShlomo E. Chazan, Sharon Gannot, Jacob Goldberger. 1-5 [doi]
- Subspace superdirective beamforming with uniform circular microphone arraysGongping Huang, Jacob Benesty, Jingdong Chen. 1-5 [doi]
- Increasing the environment-awareness of rake beamforming for directive acoustic sourcesPasi Pertilä, Alessio Brutti. 1-5 [doi]
- On pre-filtering strategies for the GCC-PHAT algorithmHong-Goo Kang, Michael Graczyk, Jan Skoglund. 1-5 [doi]
- Fast simulation method for room impulse responses based on the mirror image source assumptionJia Yan, W. Bastiaan Kleijn. 1-5 [doi]
- Performance comparison of intrusive and non-intrusive instrumental quality measures for enhanced speechAnderson R. Avila, Benjamin Cauchi, Stefan Goetze, Simon Doclo, Tiago H. Falk. 1-5 [doi]
- Binaural speech enhancement using a codebook based approachMathew Shaji Kavalekalam, Mads Græsbøll Christensen, Jesper Bünsow Boldt. 1-5 [doi]
- Anechoic phase estimation from reverberant signalsA. Belhomme, Yves Grenier, R. Badeau, E. Humbert. 1-5 [doi]
- A directionaly constrained distortionless multistage LCMV beamformerDaniel Wolff, Yaakov Buchris, Israel Cohen. 1-5 [doi]
- Solving permutation problem with a cascade combination of phase difference entropy and power spectral correlationMasahito Togami, Ryoichi Takashima, Yusuke Fujita. 1-4 [doi]
- The open-set problem in acoustic scene classificationDaniele Battaglino, Ludovick Lepauloux, Nicholas W. D. Evans. 1-5 [doi]
- An iterative method for equalization of multichannel acoustic systems robust to system identification errorsWancheng Zhang, Patrick A. Naylor. 1-5 [doi]
- Under-modelled blind system identification for time delay estimation in reverberant environmentsWei Xue, Mike Brookes, Patrick A. Naylor. 1-5 [doi]
- HRTF-based robust least-squares frequency-invariant polynomial beamformingHendrik Barfuss, Marcel Mueglich, Walter Kellermann. 1-5 [doi]
- ALE for robots! A single-channel approach to robot self-noise cancellationJalal Taghia, Dorothea Kolossa, Rainer Martin. 1-5 [doi]
- Relative impulse response estimation during doubletalk with an artificial neural network-based step size controlStefan Meier, Walter Kellermann. 1-5 [doi]
- Dual-microphone phase-difference-based SNR estimation with applications to speech enhancementFrédéric Mustière, Renato Nakagawa, Kamil K. Wójcicki, Ivo Merks, Tao Zhang. 1-5 [doi]
- A robust data-independent near-field beamformer for linear microphone arraysF. Borra, Lucio Bianchi, Fabio Antonacci, Stefano Tubaro, Augusto Sarti. 1-5 [doi]
- Robust TDOA-based joint source and microphone localization in a reverberant environment using medians of acceptable recovered TOAsTrung-Kien Le, Nobutaka Ono. 1-5 [doi]
- Sparseness-based multichannel nonnegative matrix factorization for blind source separationTakuya Higuchi, Takuya Yoshioka, Tomohiro Nakatani. 1-5 [doi]
- Speech enhancement using a microphone array mounted on an unmanned aerial vehicleYusuke Hioka, Michael Kingan, Gian Schmid, Karl A. Stol. 1-5 [doi]
- A computationally cheaper method for blind speech separation based on AuxIVA and incomplete demixing transformJakub Janský, Zbynek Koldovský, Nobutaka Ono. 1-5 [doi]
- Extraction of exterior field from a mixed sound field for 2D height-invariant sound propagationAbdullah Fahim, Prasanga N. Samarasinghe, Thushara D. Abhayapala. 1-5 [doi]
- Cue-preserving MMSE filter for binaural speech enhancementGerald Enzner, Masoumeh Azarpour, Jan Siska. 1-5 [doi]
- Synthesis of device-independent noise corpora for speech quality assessmentHannes Gamper, Lyle Corbin, David Johnston, Ivan J. Tashev. 1-5 [doi]
- Variable step-size diffusion proportionate affine projection algorithmJuan Shi, Jingen Ni, Xiaoping Chen. 1-4 [doi]
- An intelligibility metric based on a simple model of speech communicationSteven Van Kuyk, W. Bastiaan Kleijn, Richard C. Hendriks. 1-5 [doi]
- Voice activity detection based on statistical likelihood ratio with adaptive thresholdingXiaofei Li, Radu Horaud, Laurent Girin, Sharon Gannot. 1-5 [doi]
- Bi-magnitude processing framework for nonlinear acoustic echo cancellation on Android devicesYiteng Arden Huang, Jan Skoglund, Alejandro Luebs. 1-5 [doi]
- First-order differential microphone arrays from a time-domain broadband perspectiveYaakov Buchris, Israel Cohen, Jacob Benesty. 1-5 [doi]
- Recursive implementations of informed spatial filtersMaja Taseska, Reza Varzandeh, Emanuel A. P. Habets. 1-5 [doi]
- A real-time noise energy estimation methodYaodu Wei, Li Liu, Lizhong Wang. 1-4 [doi]
- Noise reduction using independent vector analysis and noise cancellation for a hose-shaped rescue robotMasaru Ishimura, Shoji Makino, Takeshi Yamada, Nobutaka Ono, Hiroshi Saruwatari. 1-5 [doi]
- Application of neural network to source PSD estimation for wiener filter based array sound source enhancementTomoko Kawase, Kenta Niwa, Kazunori Kobayashi, Yusuke Hioka. 1-5 [doi]
- Multi-speaker DOA estimation in reverberation conditions using expectation-maximizationOfer Schwartz, Yuval Dorfan, Emanuel A. P. Habets, Sharon Gannot. 1-5 [doi]
- Assessing the segmental contribution to the non-intrusive intelligibility prediction of noise-suppressed speechLei Wang, Fei Chen. 1-4 [doi]
- A modified a priori SER for acoustic echo suppression using wiener filterYing Tong, Yaping Gu. 1-4 [doi]
- Spherical microphone array post-filtering for reverberation suppression using isotropic beamformingsYuhei Yamamoto, Yoichi Haneda. 1-5 [doi]
- Inter-channel coherence reduction method for stereophonic acoustic echo cancellationTomas Gänsler, Eric J. Diethorn. 1-5 [doi]
- Hammerstein model-based nonlinear echo cancelation using a cascade of neural network and adaptive linear filterJirí Málek, Zbynek Koldovský. 1-5 [doi]
- A tunable beamformer for robust superdirective beamformingReuven Berkun, Israel Cohen, Jacob Benesty. 1-5 [doi]
- Acoustic feedback cancellation for a multi-microphone earpiece based on a null-steering beamformerHenning F. Schepker, Linh T. T. Tran, Sven Nordholm, Simon Doclo. 1-5 [doi]
- Head-orientation compensation with video-informed single channel speech enhancementSoumitro Chakrabarty, Deepth Pilakeezhu, Emanuel A. P. Habets. 1-5 [doi]
- Perceptual improvement of deep neural networks for monaural speech enhancementWei Han, Xiongwei Zhang, Meng Sun, Wenhua Shi, Xushan Chen, Yonggang Hu. 1-5 [doi]
- Efficient initialization for nonnegative matrix factorization based on nonnegative independent component analysisDaichi Kitamura, Nobutaka Ono. 1-5 [doi]
- Mask estimate through Itakura-Saito nonnegative RPCA for speech enhancementGang Min, Xiongwei Zhang, Xia Zou, Meng Sun. 1-5 [doi]
- Improved nonnegative adaptive filtering algorithmsKai Zhao, Jingen Ni, Xiaoping Chen. 1-4 [doi]
- Statistical analysis and improvement of coherent-to-diffuse power ratio estimators for dereverberationChengshi Zheng, Xiaodong Li, Andreas Schwarz, Walter Kellermann. 1-5 [doi]
- Oracle performance investigation of the ideal masksZiteng Wang, Xiaofei Wang, Xu Li, Qiang Fu, Yonghong Yan 0002. 1-5 [doi]
- Partitioned block frequency domain Kalman filter for multi-channel linear prediction based blind speech dereverberationT. Bietzen, Ann Spriet, Wouter Tirry, Simon Doclo, Marc Moonen, Toon van Waterschoot. 1-5 [doi]