| 1687 | -- | 0 | Li Deng. Farewell editorial: keeping up the momentum of innovations |
| 1688 | -- | 1700 | Sree Harsha Yella, Hervé Bourlard. Overlapping speech detection using long-term conversational features for speaker diarization in meeting room conversations |
| 1701 | -- | 1712 | Ravi K. Chivukula, Yuriy A. Reznik, Yanyan Hu, Venkat Devarajan, Mythreya Jayendra-Lakshman. Fast algorithms for low-delay TDAC filterbanks in MPEG-4 AAC-ELD |
| 1713 | -- | 1725 | Shaofei Xue, Ossama Abdel Hamid, Hui Jiang 0001, Li-Rong Dai, Qingfeng Liu. Fast adaptation of deep neural network based on discriminant codes for speech recognition |
| 1726 | -- | 1737 | Matthew E. P. Davies, Philippe Hamel, Kazuyoshi Yoshii, Masataka Goto. AutoMashUpper: automatic creation of multi-song music mashups |
| 1738 | -- | 1749 | Chao Weng, David L. Thomson, Patrick Haffner, Biing-Hwang Juang. Latent semantic rational kernels for topic spotting on conversational speech |
| 1750 | -- | 1764 | Neil Wachowski, Mahmood R. Azimi-Sadjadi. Detection and classification of nonstationary transient signals using sparse approximations and Bayesian networks |
| 1765 | -- | 1776 | Graham Percival, George Tzanetakis. Streamlined tempo estimation based on autocorrelation and cross-correlation with pulses |
| 1777 | -- | 1791 | Annea Barkefors, Mikael Sternad, Lars-Johan Brännmark. Design and analysis of linear quadratic Gaussian feedforward controllers for active noise control |
| 1792 | -- | 1802 | Maximo Cobos, Juan José Pérez Solano, Santiago Felici-Castell, Jaume Segura, Juan M. Navarro. Cumulative-sum-based localization of sound events in low-cost wireless acoustic sensor networks |
| 1803 | -- | 1814 | Vladimir Tourbabin, Boaz Rafaely. Theoretical framework for the optimization of microphone array configuration for humanoid robot audition |
| 1815 | -- | 1824 | Yuriy V. Zakharov, Vitor H. Nascimento. Sliding-window RLS low-cost implementation of proportionate affine projection algorithms |
| 1825 | -- | 1832 | Stefano D'Angelo, Vesa Välimäki. Generalized Moog ladder filter: part I-linear analysis and parameterization |
| 1833 | -- | 1848 | Na Yang, He Ba, Weiyang Cai, Ilker Demirkol, Wendi Heinzelman. BaNa: a noise resilient fundamental frequency detection algorithm for speech and music |
| 1849 | -- | 1858 | Yuxuan Wang, Arun Narayanan, DeLiang Wang. On training targets for supervised speech separation |
| 1859 | -- | 1872 | Ling-Hui Chen, Zhen-Hua Ling, Li-juan Liu, Li-Rong Dai. Voice conversion using deep neural networks with layer-wise generative training |
| 1873 | -- | 1883 | Stefano D'Angelo, Vesa Välimäki. Generalized Moog ladder filter: part II-explicit nonlinear model through a novel delay-free loop implementation method |
| 1884 | -- | 1893 | Zafar Rafii, Zhiyao Duan, Bryan Pardo. Combining rhythm-based and pitch-based methods for background and melody separation |
| 1894 | -- | 1904 | Jussi Rämö, Vesa Välimäki, Balázs Bank. High-precision parallel graphic equalizer |
| 1905 | -- | 1917 | Yannis Panagakis, Constantine Kotropoulos, Gonzalo R. Arce. Music genre classification via joint sparse low-rank representation of audio features |
| 1918 | -- | 1930 | Akira Maezawa, Katsutoshi Itoyama, Kazuyoshi Yoshii, Hiroshi G. Okuno. Nonparametric Bayesian dereverberation of power spectrograms based on infinite-order autoregressive processes |
| 1931 | -- | 1940 | Martin Krawczyk, Timo Gerkmann. STFT phase reconstruction in voiced speech for an improved single-channel speech enhancement |
| 1941 | -- | 1950 | Vahid Khanagha, Khalid Daoudi, Hussein M. Yahia. Detection of glottal closure instants based on the microcanonical multiscale formalism |
| 1951 | -- | 1964 | A. Venturini, L. Zão, R. Coelho. On speech features fusion, α-integration Gaussian modeling and multi-style training for noise robust speaker classification |
| 1965 | -- | 1977 | Peter Foster, Matthias Mauch, Simon Dixon. Sequential complexity as a descriptor for musical similarity |
| 1978 | -- | 1992 | Gang Liu, John H. L. Hansen. An investigation into back-end advancements for speaker recognition in multi-session and noisy enrollment scenarios |
| 1993 | -- | 2002 | Jitong Chen, Yuxuan Wang, DeLiang Wang. A feature study for classification-based speech separation at low signal-to-noise ratios |
| 2003 | -- | 2011 | Jelle Van Mourik, Damian Murphy. Explicit higher-order FDTD schemes for 3D room acoustic simulation |
| 2012 | -- | 2024 | Pei Chee Yong, Sven Nordholm, Hai Huyen Dam. Effective binaural multi-channel processing algorithm for improved environmental presence |
| 2025 | -- | 2033 | Austin Chen, Mark A. Hasegawa-Johnson. Mixed stereo audio classification using a stereo-input mixed-to-panned level feature |
| 2034 | -- | 2047 | Gongping Huang, Jacob Benesty, Tao Long, Jingdong Chen. A family of maximum SNR filters for noise reduction |
| 2048 | -- | 2058 | Su Yan, Xiaojun Wan. SRRank: leveraging semantic roles for extractive multi-document summarization |
| 2059 | -- | 2073 | Hideyuki Tachibana, Nobutaka Ono, Hirokazu Kameoka, Shigeki Sagayama. Harmonic/percussive sound separation based on anisotropic smoothness of spectrograms |
| 2074 | -- | 2086 | Jose Manuel Gil-Cacho, Toon van Waterschoot, Marc Moonen, Søren Holdt Jensen. A frequency-domain adaptive filter (FDAF) prediction error method (PEM) framework for double-talk-robust acoustic echo cancellation |
| 2087 | -- | 2100 | Qi Wang, W. L. Woo, S. S. Dlay. Informed single-channel speech separation using HMM-GMM user-generated exemplar source |
| 2101 | -- | 2111 | Daniel Erro, Tudor-Catalin Zorila, Yannis Stylianou. Enhancing the intelligibility of statistically generated synthetic speech by means of noise-independent modifications |
| 2112 | -- | 2121 | Yi Jiang, DeLiang Wang, Runsheng Liu, Zhenming Feng. Binaural classification for reverberant speech segregation using deep neural networks |
| 2122 | -- | 2132 | Li Su, Hsin-Ming Lin, Yi-Hsuan Yang. Sparse modeling of magnitude and phase-derived spectra for playing technique classification |
| 2133 | -- | 2145 | Vinod Veera Reddy, Andy W. H. Khong, Boon Poh Ng. Unambiguous speech DOA estimation under spatial aliasing conditions |
| 2146 | -- | 2157 | Amir Mohammadi, Seyyed Saeed Sarfjoo, Cenk Demiroglu. Eigenvoice speaker adaptation with minimal data for statistical speech synthesis systems using a MAP approach and nearest-neighbors |
| 2158 | -- | 2168 | Kun Han, DeLiang Wang. Neural network based pitch tracking in very noisy speech |
| 2169 | -- | 2181 | Yongsheng Mu, Peifeng Ji, Wei Ji, Ming Wu, Jun Yang. Modeling and compensation for the distortion of parametric loudspeakers using a one-dimension Volterra filter |
| 2182 | -- | 2196 | Oliver Thiergart, Maja Taseska, Emanuel A. P. Habets. An informed parametric spatial filter based on instantaneous direction-of-arrival estimates |
| 2197 | -- | 2206 | João Felipe Santos, Tiago H. Falk. Updating the SRMR-CI metric for improved intelligibility prediction for cochlear implant users |
| 2207 | -- | 2217 | Seon-Man Kim, Hong Kook Kim. Direction-of-arrival based SNR estimation for dual-microphone speech enhancement |
| 2218 | -- | 2232 | Takuma Otsuka, Katsuhiko Ishiguro, Takuya Yoshioka, Hiroshi Sawada, Hiroshi G. Okuno. Multichannel sound source dereverberation and separation for arbitrary number of sources based on Bayesian nonparametrics |
| 2233 | -- | 2243 | Johannes Traa, Paris Smaragdis. Multichannel source separation and tracking with RANSAC and directional statistics |
| 2244 | -- | 2255 | Weifeng Li, Longbiao Wang, Yicong Zhou, John Dines, Mathew Magimai-Doss, Hervé Bourlard, Qingmin Liao. Feature mapping of multiple beamformed sources for robust overlapping speech recognition using a microphone array |
| 2256 | -- | 0 | Yi FanChiang, Cheng-Wen Wei, Yi-Le Meng, Yu-Wen Lin, Shyh-Jye Jou, Tian-Sheuan Chang. Correction to "Low complexity formant estimation adaptive feedback cancellation for hearing aids using pitch based processing" |