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ICASSP 2011 Speech Acoustics

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ICASSP
2011
IEEE

921views Signal Processing» more ICASSP 2011»

Non-flat clustering whith alpha-divergences

13 years 9 months ago

Download www.lix.polytechnique.fr

The scope of the well-known k-means algorithm has been broadly extended with some recent results: first, the k- means++ initialization method gives some approximation guarantees...

Olivier Schwander, Frank Nielsen

posted by schwander

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Id	Views	Title	Status

1	921	Non-flat clustering whith alpha-divergences
2	706	Fast single image fog removal using edge-preserving smoothing	claim
3	468	Single image dehazing based on contrast enhancement	claim
4	446	Searching in one billion vectors: re-rank with source coding
5	419	Automatic video annotation via Hierarchical Topic Trajectory Model considering cross-modal correlations	claim
6	354	Time domain synchronization and decoding of P1 symbol in DVB-T2	claim
7	350	Multiple-Measurement Vector model and its application to Through-the-Wall Radar Imaging	claim
8	347	An introduction to consistent graphs and their signal processing applications	claim
9	345	Detecting moving objects from dynamic background with shadow removal	claim
10	343	Auto-regressive model based error concealment scheme for stereoscopic video coding	claim
11	331	Video anomaly recovery from compressed spectral imaging	claim
12	327	A constrained matching pursuit approach to audio declipping	claim
13	326	On the effectiveness of the Dark Channel Prior for single image dehazing by approximating with minimum volume ellipsoids	claim
14	326	Extensions of recurrent neural network language model	claim
15	323	High dynamic range (HDR) imaging by gradient domain fusion	claim
16	322	Multi image super resolution using compressed sensing	claim
17	320	No-reference bit stream model for video quality assessment of h.264/AVC video based on packet loss visibility	claim
18	315	Joint estimation of channel and carrier frequency offset from the emitter, in an uplink OFDMA system	claim
19	314	Region-based image fusion using a combinatory Chebyshev-ICA method	claim
20	314	Superpixel-based object class segmentation using conditional random fields	claim
21	314	On the ergodic capacity of jointly-correlated rician Fading MIMO channels	claim
22	312	Feature selection through gravitational search algorithm	claim
23	308	A novel vector quantization-based video summarization method using independent component analysis mixture model	claim
24	304	A multi-exposure image fusion algorithm without ghost effect	claim
25	303	A new data hiding method using angle quantization index modulation in gradient domain	claim
26	296	Smart grid monitoring for intrusion and fault detection with new locally optimum testing procedures	claim
27	294	An uniqueness condition for the 4-way CANDECOMP/PARAFAC model with collinear loadings in three modes	claim
28	293	Ambiguity functions of compressively sensed and processed radar waveforms	claim
29	292	Compressive sensing in through-the-wall radar imaging	claim
30	291	RRED indices: Reduced reference entropic differencing framework for image quality assessment	claim
31	291	Rotation invariant feature extraction from 3-D acceleration signals	claim
32	291	Combining generic and class-specific codebooks for object categorization and detection	claim
33	291	Semi-automatic 2D to 3D image conversion using a hybrid Random Walks and graph cuts based approach	claim
34	291	Downsampling graphs using spectral theory	claim
35	290	Improved model-based spectral compressive sensing via nested least squares	claim
36	288	Learning a discriminative visual codebook using homonym scheme	claim
37	287	Signal processing in medical imaging and image-guided intervention	claim
38	286	Analog joint source-channel coding in Rayleigh fading channels	claim
39	286	A hybrid compressed sensing algorithm for sparse channel estimation in MIMO OFDM systems	claim
40	286	Adaptation of source-specific dictionaries in Non-Negative Matrix Factorization for source separation	claim
41	285	Compressed sensing based method for ECG compression	claim
42	283	An adaptive bayesian clustering and multivariate region merging based technique for efficient segmentation of color images	claim
43	283	WinDSK8: A user interface for the OMAP-L138 DSP board	claim
44	282	Efficient block-division model for robust multiple object tracking	claim
45	281	Multiple kernel nonnegative matrix factorization	claim
46	279	Using the kernel trick in compressive sensing: Accurate signal recovery from fewer measurements	claim
47	279	Automatic target classification in SAR images using MPCA	claim
48	279	Score fusion and calibration in multiple language detectors with large performance variation	claim
49	277	On the identification of parametric underspread linear systems	claim
50	276	Bayesian sensing hidden Markov models for speech recognition	claim
51	275	Adaptive compressive sensing and processing for radar tracking	claim
52	274	Directional Color Filter Array interpolation based on multiscale color gradients	claim
53	274	Sparsity-based defect pixel compensation for arbitrary camera raw images	claim
54	273	A wavenumber-fitting extrapolation method for FFT-based near-field acoustic holography using microphone array	claim
55	272	3D image geo-registration using vision-based modeling	claim
56	271	Uncover cooperative gene regulations by microRNAs and transcription factors in glioblastoma using a nonnegative hybrid factor model	claim
57	271	Source localization using time difference of arrival within a sparse representation framework	claim
58	271	Convergence results in distributed Kalman filtering	claim
59	270	Content preference estimation in online social networks: Message passing versus sparse reconstruction on graphs	claim
60	270	Alignment of cubic-panorama image datasets using epipolar geometry	claim
61	269	Automatic Language Identification in music videos with low level audio and visual features	claim
62	267	Rapid image retrieval for mobile location recognition	claim
63	265	Sparse graphical modeling of piecewise-stationary time series	claim
64	263	An automated singing evaluation method for Karaoke systems	claim
65	262	Hierarchical Latent Dirichlet Allocation models for realistic action recognition	claim
66	262	Cooperative spectrum sensing based on matrix rank minimization	claim
67	261	A novel fast canonical-signed-digit conversion technique for multiplication	claim
68	261	Combining Haar Feature and skin color based classifiers for face detection	claim
69	261	Feature selection based on Multiple Kernel Learning for single-channel sound source localization using the acoustic transfer function	claim
70	261	Statistical analysis of multi-channel detection using data from airborne AESA radar	claim
71	261	Cosparse analysis modeling - uniqueness and algorithms	claim
72	260	Azimuth-elevation direction finding using power measurements from single antenna	claim
73	259	An ALPS view of sparse recovery	claim
74	259	HNM-based MFCC+F0 extractor applied to statistical speech synthesis	claim
75	259	The value of redundant measurement in compressed sensing	claim
76	259	Online detection of vocal Listener Responses with maximum latency constraints	claim
77	258	CYCLOP: A stereo color image quality assessment metric	claim
78	257	Dual domain method for single image dehazing and enhancing	claim
79	256	A convex approximation approach to weighted sum rate maximization of multiuser MISO interference channel under outage constraints	claim
80	256	Resource allocation for OFDMA cognitive radios under channel uncertainty	claim
81	256	Loquendo - Politecnico di Torino's 2010 NIST speaker recognition evaluation system	claim
82	255	Geometric multichannel common signal separation with application to music and effects extraction from film soundtracks	claim
83	255	Beyond bag of words: Combining generative and discriminative models for natural scene categorization	claim
84	255	A single-channel noise reduction MVDR filter	claim
85	255	Exploring nuisance attribute projection and score normalization for GLDS-SVM based automatic mispronunciation detection method	claim
86	255	A region based approach to background modeling in a wavelet multi-resolution framework	claim
87	254	Sparse decomposition of transformation-invariant signals with continuous basis pursuit	claim
88	254	Adaptive beam tracking for interference alignment in time-varying MIMO interference channels: Conjugate gradient approach	claim
89	254	Evaluation of adaptive blind SIMO identification in terms of a normalized filter-projection misalignment	claim
90	254	Video object tracking with differential Structural SIMilarity index	claim
91	254	Generic object recognition using automatic region extraction and dimensional feature integration utilizing multiple kernel learning	claim
92	252	A complete ensemble empirical mode decomposition with adaptive noise	claim
93	252	Efficient stereo-to-multiview synthesis	claim
94	251	Efficiently computing private recommendations	claim
95	251	Optical MEMS image enhancement with sparse signal representation	claim
96	250	Dirichlet Mixture Models of neural net posteriors for HMM-based speech recognition	claim
97	250	Effective background data selection in SVM speaker recognition for unseen test environment: More is not always better	claim
98	249	Building HMM based unit-selection speech synthesis system using synthetic speech naturalness evaluation score	claim
99	249	BEMD for expression transformation in face recognition	claim
100	249	Robust video object tracking based on multiple kernels with projected gradients	claim
101	249	A mutual information based approach for evaluating the quality of clustering	claim
102	248	Robust hyperspectral signal unmixing in the presence of correlated noise	claim
103	248	Discriminatively trained Probabilistic Linear Discriminant Analysis for speaker verification	claim
104	248	Analysis of phone confusion in EMG-based speech recognition	claim
105	248	Low-complexity predictive lossy compression of hyperspectral and ultraspectral images	claim
106	247	Compensation of partly reliable components for band-limited speech recognition with missing data techniques	claim
107	247	Maximum margin structure learning of Bayesian network classifiers	claim
108	247	Down-scaling interlace video from H.264/AVC intra-coded bit stream	claim
109	247	Optimal wireless networks based on local channel state information	claim
110	247	Spatially sparsed Common Spatial Pattern to improve BCI performance	claim
111	246	Data aided phase tracking and symbol detection for CPM in frequency-flat fading channel	claim
112	246	Gaussian mixture modeling for source localization	claim
113	245	Multi-cue based multi-target tracking using online random forests	claim
114	245	Bayesian Compressive Sensing for clustered sparse signals	claim
115	245	Error detection scheme based on fragile watermarking for H.264/AVC	claim
116	245	Acoustic data sharing for Afghan and Persian languages	claim
117	244	An investigation of subspace modeling for phonetic and speaker variability in automatic speech recognition	claim
118	244	Bounded gradient projection methods for sparse signal recovery	claim
119	243	Feature-based global motion estimation using the Helmholtz principle	claim
120	243	Theoretical analyses on a class of nested RKHS's	claim
121	243	Adaptive appearance learning for visual object tracking	claim
122	242	Rate control for PSD limited multiple access systems through linear programming	claim
123	242	SSIM-inspired image denoising using sparse representations	claim
124	242	Single-frame-based rain removal via image decomposition	claim
125	242	Clustering of bootstrapped acoustic model with full covariance	claim
126	241	Edge oriented directional Color Filter Array interpolation	claim
127	241	Fusing shape and texture information for facial age estimation	claim
128	241	RSS-based node localization in the presence of attenuating objects	claim
129	241	Joint source-channel rate control for pixel-domain distributed video coding	claim
130	241	Contour-based hidden Markov model to segment 2D ultrasound images	claim
131	240	Stochastic behavior analysis of the Gaussian Kernel Least Mean Square algorithm	claim
132	240	Unsupervised extraction of audio-visual objects	claim
133	240	Smart meter privacy using a rechargeable battery: Minimizing the rate of information leakage	claim
134	240	Occlusion boundary detection using an online learning framework	claim
135	240	An acoustically-motivated spatial prior for under-determined reverberant source separation	claim
136	239	Structural MAP adaptation in GMM-supervector based speaker recognition	claim
137	239	Coordinated user scheduling in the multi-cell MIMO downlink	claim
138	239	A game theoretic approach to eavesdropper cooperation in MISO wireless networks	claim
139	239	Relevance language modeling for speech recognition	claim
140	239	Analysis of human footsteps utilizing multi-axial seismic fusion	claim
141	239	Compact support kernels based time-frequency distributions: Performance evaluation	claim
142	239	Biological pathway inference using manifold embedding	claim
143	239	Joint Bayesian removal of impulse and background noise	claim
144	238	A revisit to MRF-based depth map super-resolution and enhancement	claim
145	238	Two dimensional nested arrays on lattices	claim
146	238	A cross-layer optimization for energy-efficient MAC protocol with delay and rate constraints	claim
147	238	A robust lip tracking algorithm using localized color active contours and deformable models	claim
148	238	A cochlear neuron based robust feature for speaker recognition	claim
149	238	Audio signal classification with temporal envelopes	claim
150	237	A regularization framework for mobile social network analysis	claim
151	237	Robust adaptive beamforming based on jointly estimating covariance matrix and steering vector	claim
152	237	In-class demonstrations with a portable laboratory for teaching DSP to Computer Engineering majors	claim
153	237	Audio Signal Representations for Factorization in the Sparse Domain	claim
154	237	Fast adaptive variational sparse Bayesian learning with automatic relevance determination	claim
155	236	A method to infer emotions from facial Action Units	claim
156	236	Distributed LCMV beamforming in wireless sensor networks with node-specific desired signals	claim
157	235	Facial expression recognition using ensemble of classifiers	claim
158	235	A combined linear programming-maximum likelihood approach to radial velocity data analysis for extrasolar planet detection	claim
159	235	Temporally coherent luminance-to-luma mapping for high dynamic range video coding with H.264/AVC	claim
160	234	Acceleration of adaptive proximal forward-backward splitting method and its application to sparse system identification	claim
161	234	Basis pursuit for spectrum cartography	claim
162	233	A fully automated 2-DE gel image analysis pipeline for high throughput proteomics	claim
163	232	Automatic identification of speaker role and agreement/disagreement in broadcast conversation	claim
164	232	Belief theoretic methods for soft and hard data fusion	claim
165	232	Speaker recognition using multiple kernel learning based on conditional entropy minimization	claim
166	232	Sparse common spatial patterns in brain computer interface applications	claim
167	231	Optimum multi-user detection by nonsmooth optimization	claim
168	231	An iterative least-squares technique for dereverberation	claim
169	230	Fast estimation of the state of the power grid using synchronized phasor measurements	claim
170	229	A point target reference spectrum for general bistatic SAR processing	claim
171	229	Learning and inference algorithms for partially observed structured switching vector autoregressive models	claim
172	229	Deep neural networks for acoustic emotion recognition: Raising the benchmarks	claim
173	228	Multiple instance tracking based on hierarchical maximizing bag's margin boosting	claim
174	228	Estimation and dynamic updating of time-varying signals with sparse variations	claim
175	228	Fast algorithms for Iterative Adaptive Approach spectral estimation techniques	claim
176	228	Image editing based on Sparse Matrix-Vector multiplication	claim
177	227	Accurate transcription of broadcast news speech using multiple noisy transcribers and unsupervised reliability metrics	claim
178	227	Comparing multilayer perceptron to Deep Belief Network Tandem features for robust ASR	claim
179	227	Cheap semidefinite relaxation MIMO detection using row-by-row block coordinate descent	claim
180	227	Investigation of acoustic units for LVCSR systems	claim
181	227	Proximal splitting methods for depth estimation	claim
182	226	Infinite-state spectrum model for music signal analysis	claim
183	226	Analysis of unknown velocity and target off the grid problems in compressive sensing based subsurface imaging	claim
184	226	A fast transform for acoustic imaging with separable arrays	claim
185	226	Automatic speech recognition using Hidden Conditional Neural Fields	claim
186	226	A Bernoulli-Gaussian model for gene factor analysis	claim
187	226	A CLT on the SINR of the diagonally loaded Capon/MVDR beamformer	claim
188	225	Look ahead orthogonal matching pursuit	claim
189	225	L0 sparse graphical modeling	claim
190	225	Secrecy capacity and secure outage performance for rayleigh fading simo channel	claim
191	224	Dictionary learning of convolved signals	claim
192	224	Random finite set for data association in multiple camera tracking	claim
193	224	Incoherent color frames for compressive demosaicing	claim
194	223	Compressive power spectral density estimation	claim
195	223	An extension of the ICA model using latent variables	claim
196	223	Collaborative sources identification in mixed signals via hierarchical sparse modeling	claim
197	222	Sliding window greedy RLS for sparse filters	claim
198	222	Robust changepoint detection based on multivariate rank statistics	claim
199	222	Discriminatively estimated discrete, parametric and smoothed-discrete duration models for speech recognition	claim
200	221	Exploiting active-learning strategies for annotating prosodic events with limited labeled data	claim
201	221	A segmentation method for textured images based on the maximum posterior mode criterion	claim
202	221	Efficient context adaptive entropy coding for real-time applications	claim
203	221	Denoising of image patches via sparse representations with learned statistical dependencies	claim
204	220	Distributed one bit feedback extended orthogonal space time coding based on selection of cyclic rotation for cooperative relay networks	claim
205	220	A 3-layer coding scheme for biometry template protection based on spectral minutiae	claim
206	220	The HKCUPU system for the NIST 2010 speaker recognition evaluation	claim
207	220	A distance-based slice interleaving scheme for robust video transmission over error-prone networks	claim
208	219	A segment-based image saliency detection	claim
209	219	Active noise control in headsets: A new approach for broadband feedback ANC	claim
210	219	Video thumbnail extraction using video time density function and independent component analysis mixture model	claim
211	219	Analysis of rate constraints for MWF-based noise reduction in acoustic sensor networks	claim
212	218	Audio source separation by basis function adaptation	claim
213	218	Adjugate pairs of sparse arrays for sampling two dimensional signals	claim
214	218	Denoising sparse noise via online dictionary learning	claim
215	218	Comparative threshold performance study for conditional and unconditional direction-of-arrival estimation	claim
216	217	Efficient out-of-vocabulary term detection by n-gram array indices with distance from a syllable lattice	claim
217	217	User selection schemes for maximizing throughput of multiuser MIMO systems using Zero Forcing Beamforming	claim
218	217	Joint bandwidth and power allocation in cognitive radio networks under fading channels	claim
219	217	A visual attention model combining top-down and bottom-up mechanisms for salient object detection	claim
220	217	Integrating frame-based and segment-based dynamic time warping for unsupervised spoken term detection with spoken queries	claim
221	216	Gain-robust multi-pitch tracking using sparse nonnegative matrix factorization	claim
222	216	Consensus-based distributed particle filtering algorithms for cooperative blind equalization in receiver networks	claim
223	215	Generalized Restricted Isometry Property for alpha-stable random projections	claim
224	215	Using morpheme and syllable based sub-words for polish LVCSR	claim
225	215	Discriminative duration modeling for speech recognition with segmental conditional random fields	claim
226	215	Image compression using learned dictionaries by RLS-DLA and compared with K-SVD	claim
227	215	MLP based phoneme detectors for Automatic Speech Recognition	claim
228	215	Detecting low-rate periodic events in Internet traffic using renewal theory	claim
229	215	Topic-sensitive interactive image object retrieval with noise-proof relevance feedback	claim
230	214	OpenBliSSART: Design and evaluation of a research toolkit for Blind Source Separation in Audio Recognition Tasks	claim
231	214	Unsupervised vocabulary discovery using non-negative matrix factorization with graph regularization	claim
232	214	Approximate nearest-subspace representations for sound mixtures	claim
233	214	Approximated kernel density estimation for multiple TDOA detection	claim
234	214	A new stochastic image model based on Markov random fields and its application to texture modeling	claim
235	214	The Centered Discrete Fourier Transform and a parallel implementation of the FFT	claim
236	214	Itakura-Saito nonnegative matrix factorization with group sparsity	claim
237	213	An evaluation of alaryngeal speech enhancement methods based on voice conversion techniques	claim
238	212	Co-clustering as multilinear decomposition with sparse latent factors	claim
239	212	Two dimensional non-separable adaptive directional lifting structure of discrete wavelet transform	claim
240	212	Space-time adaptive processing for range-folded spread-Doppler radar clutter mitigation	claim
241	212	Shannon meets Nyquist: Capacity limits of sampled analog channels	claim
242	212	Analog antenna combining in multiuser OFDM systems: Beamforming design and power allocation	claim
243	212	On 2D localization of reflectors using robust beamforming techniques	claim
244	211	Filter design with secrecy constraints: The multiple-input multiple-output Gaussian wiretap channel with zero forcing receive filters	claim
245	211	Clustering NMF basis functions using Shifted NMF for monaural sound source separation	claim
246	211	Nonparametric Bayesian feature selection for multi-task learning	claim
247	211	Delta-spectral cepstral coefficients for robust speech recognition	claim
248	211	Speaker verification using sparse representation classification	claim
249	210	Applications of short space-time fourier analysis in digital acoustics	claim
250	210	Can one hear the shape of a room: The 2-D polygonal case	claim
251	210	Audiovisual classification of vocal outbursts in human conversation using Long-Short-Term Memory networks	claim
252	210	Informed secure watermarking using optimal transport	claim
253	210	Anti-aliasing filter for subpixel down-sampling based on frequency analysis	claim
254	210	Deep Belief Networks using discriminative features for phone recognition	claim
255	210	MIMO radar diversity with Neyman-Pearson signal detection in non-Gaussian circumstance with non-orthogonal waveforms	claim
256	209	Analysis of the security of linear blinding techniques from an information theoretical point of view	claim
257	209	On autocorrelation-based multiantenna spectrum sensing for cognitive radios in unknown noise	claim
258	209	Multiple speaker tracking using a microphone array by combining auditory processing and a gaussian mixture cardinalized probability hypothesis density filter	claim
259	209	EM-style optimization of hidden conditional random fields for grapheme-to-phoneme conversion	claim
260	209	A robust feature extraction algorithm based on class-Modular Image Principal Component Analysis for face verification	claim
261	209	Leveraging the Web for automatically generating indexable and browsable keywords for speech files	claim
262	209	Correlation matrix interpolation in Sound Source Localization for a robot	claim
263	209	Convex approximation algorithms for back-pressure power control of wireless multi-hop networks	claim
264	208	A clustering based framework for dictionary block structure identification	claim
265	208	Compensation of extrinsic variability in speaker verification systems on simulated Skype and HF channel data	claim
266	208	Combining monaural source separation with Long Short-Term Memory for increased robustness in vocalist gender recognition	claim
267	208	TLS-FOCUSS for sparse recovery with perturbed dictionary	claim
268	208	Joint algorithm-architecture optimization of CABAC to increase speed and reduce area cost	claim
269	208	A novel algorithm for occlusions and perspective effects using a 3D object process	claim
270	208	Sparsity-undersampling tradeoff of compressed sensing in the complex domain	claim
271	208	Sparse non-negative decomposition of speech power spectra for formant tracking	claim
272	207	A VAD-robust Multichannel Wiener Filter algorithm for noise reduction in hearing aids	claim
273	207	Improving acoustic event detection using generalizable visual features and multi-modality modeling	claim
274	207	Including human expertise in speaker recognition systems: report on a pilot evaluation	claim
275	206	Dense disparity estimation from linear measurements	claim
276	206	Asymmetric acoustic modeling of mixed language speech	claim
277	206	Improving melody extraction using Probabilistic Latent Component Analysis	claim
278	206	Event classification for personal photo collections	claim
279	206	Assessing the speaker recognition performance of naive listeners using mechanical turk	claim
280	206	Large vocabulary continuous speech recognition with context-dependent DBN-HMMS	claim
281	206	ECG for blind identity verification in distributed systems	claim
282	206	Enhancement of uneven lighting text image using line-based Empirical Mode Decomposition	claim
283	205	Color correction for object tracking across multiple cameras	claim
284	205	Structured Output Layer neural network language model	claim
285	204	Slice error concealment based on size-adaptive SSIM matching and motion vector outlier rejection	claim
286	204	Fixed- versus floating-point implementation of MIMO-OFDM detector	claim
287	204	Bayesian reinforcement learning for POMDP-based dialogue systems	claim
288	204	Fall detection in a smart room by using a fuzzy one class support vector machine and imperfect training data	claim
289	204	Factor graph switching portfolios under transaction costs	claim
290	204	Joint source-channel-network coding for bidirectional wireless relays	claim
291	204	Weighted and structured sparse total least-squares for perturbed compressive sampling	claim
292	203	Robust SINR-constrained MISO downlink beamforming: When is semidefinite programming relaxation tight?	claim
293	203	Gaussian Mixture Modeling of vowel durations for automated assessment of non-native speech	claim
294	203	Instantaneous phase tracking of oscillatory signals using emd and Rao-Blackwellised particle filtering	claim
295	203	Adapting acoustic and lexical models to dysarthric speech	claim
296	203	A trend estimation algorithm for singing pitch detection in musical recordings	claim
297	203	Bayesian framework and message passing for joint support and signal recovery of approximately sparse signals	claim
298	202	Entropy estimation using the principle of maximum entropy	claim
299	202	Transient acoustic signal classification using joint sparse representation	claim
300	202	Real-time voice conversion based on instantaneous harmonic parameters	claim
301	202	Incentive mechanism in wireless multicast	claim
302	202	Application control for fast adaptive error resilient H.264/AVC streaming over IP wireless networks	claim
303	202	The IBM 2009 GALE Arabic speech transcription system	claim
304	202	Real-time DVB-S2 LDPC decoding on many-core GPU accelerators	claim
305	202	Clean: A false alarm reduction method for SAR CCD	claim
306	202	Multichannel nonnegative tensor factorization with structured constraints for user-guided audio source separation	claim
307	202	A knapsack problem formulation for relay selection in secure cooperative wireless communication	claim
308	202	Bayesian integration of audio and visual information for multi-target tracking using a CB-member filter	claim
309	201	Rapid speaker adaptation with speaker adaptive training and non-negative matrix factorization	claim
310	201	Computing room acoustics with CUDA - 3D FDTD schemes with boundary losses and viscosity	claim
311	201	Recent development of discriminative training using non-uniform criteria for cross-level acoustic modeling	claim
312	201	A segment-level confidence measure for Spoken Document Retrieval	claim
313	201	A spectral approach to recursive end-to-end distortion estimation for sub-pixel motion-compensated video coding	claim
314	201	Latency-constrained low-complexity lattice reduction for MIMO-OFDM systems	claim
315	200	Operational performances of a MUSIC algorithm robust to outliers	claim
316	200	Noise robust bird song detection using syllable pattern-based hidden Markov models	claim
317	200	Array-based GNSS acquisition in the presence of colored noise	claim
318	200	Gammatone sub-band magnitude-domain dereverberation for ASR	claim
319	200	Role of nucleus based context in word-independent syllable stress classification	claim
320	200	Coding of unquantized spectrum sub-bands in superwideband audio codecs	claim
321	200	A stochastic compartmental approach to modeling and simulation of cancer spheroid formation and evolution	claim
322	199	A novel study and analysis on segmental gait sequence recognition	claim
323	199	A non-negative approach to semi-supervised separation of speech from noise with the use of temporal dynamics	claim
324	199	Low-rank matrix completion by variational sparse Bayesian learning	claim
325	199	Automatic real-time electric guitar audio transcription	claim
326	199	Rapid phonetic transcription using everyday life natural Chat Alphabet orthography for dialectal Arabic speech recognition	claim
327	199	Dynamic selection of a speech enhancement method for robust speech recognition in moving motorcycle environment	claim
328	199	Why word error rate is not a good metric for speech recognizer training for the speech translation task?	claim
329	198	Bayesian topic models for describing computer network behaviors	claim
330	198	Combination of stochastic understanding and machine translation systems for language portability of dialogue systems	claim
331	198	Joint reduce of metal and beam hardening artifacts using Multi-Energy map approach in X-ray Computed Tomography	claim
332	198	Subspace constrained LU decomposition of FMLLR for rapid adaptation	claim
333	198	Proportionate affine projection sign algorithms for sparse system identification in impulsive interference	claim
334	197	Segmentation of the carotid artery in ultrasound images using frequency-designed B-spline active contour	claim
335	197	Marker-based Hierarchical Segmentation and classification approach for hyperspectral imagery	claim
336	197	Detection of geometrically known targets in Through-the-Wall radar imaging	claim
337	197	Langevin and hessian with fisher approximation stochastic sampling for parameter estimation of structured covariance	claim
338	197	Design and implementation of cubic spline interpolation for spike sorting microsystems	claim
339	197	Predictive modeling of the spatiotemporal evolution of an environmental hazard and its sensor network implementation	claim
340	196	Modeling musical attributes to characterize ensemble recordings using rhythmic audio features	claim
341	196	Compressive Sensing for over-the-air ultrasound	claim
342	196	Spectral subtraction on real and imaginary modulation spectra	claim
343	196	Vulnerability of insens to denial of service attacks	claim
344	196	Motion-decision based spatiotemporal saliency for video sequences	claim
345	195	Authentication of fingerprint scanners	claim
346	195	Computing the nonnegative 3-way tensor factorization using Tikhonov regularization	claim
347	195	Destination-aware target tracking via syntactic signal processing	claim
348	195	cROVER: Improving ROVER using automatic error detection	claim
349	195	Optimal transmission strategies for channel capture mitigation in Cognitive Radio Networks	claim
350	195	Real-time conjugate gradients for online fMRI classification	claim
351	195	Motion vector recovery with Gaussian Process Regression	claim
352	194	A partial least squares framework for speaker recognition	claim
353	194	Robust secondary multicast transmit beamforming for cognitive radio networks under imperfect channel state information	claim
354	194	Phoneme selective speech enhancement using the generalized parametric spectral subtraction estimator	claim
355	194	Interference alignment in MIMO cellular networks	claim
356	194	Rao-Blackwellized particle filter for Gaussian mixture models and application to visual tracking	claim
357	194	Keypoint-based near-duplicate images detection using affine invariant feature and color matching	claim
358	194	Calibration and weight of the evidence by human listeners. The ATVS-UAM submission to NIST HUMAN-aided speaker recognition 2010	claim
359	193	Adaptive harmonic time-frequency decomposition of audio using shift-invariant PLCA	claim
360	193	Artificial bandwidth extension of spectral envelope with temporal clustering	claim
361	193	How to perform texture recognition from stochastic modeling in the wavelet domain	claim
362	193	Energy-optimized high performance FFT processor	claim
363	193	Characterization of three-dimensional data with multidimensional deformable models based on B-splines in the fourier domain	claim
364	193	Compressive tracking of doubly selective channels in multicarrier systems based on sequential delay-Doppler sparsity	claim
365	193	Real time speaker localization and detection system for camera steering in multiparticipant videoconferencing environments	claim
366	193	Power allocation optimization in OFDM-based cognitive radios based on sensing information	claim
367	193	Compressive sampling with a successive approximation ADC architecture	claim
368	193	Capacity maximization for distributed beamforming in one- and bi-directional relay networks	claim
369	193	Optimized edge appearance probability for cooperative localization based on tree-reweighted nonparametric belief propagation	claim
370	193	A novel tracking analysis of the Normalized Least Mean Fourth algorithm	claim
371	192	Directional descriptors using zernike moment phases for object orientation estimation in underwater sonar images	claim
372	192	Discriminative Training for direct minimization of deletion, insertion and substitution errors	claim
373	192	Additive character sequences with small alphabets for compressed sensing matrices	claim
374	192	Blind phase recovery in QAM communication systems using characteristic function	claim
375	192	Robust distributed detection, localization, and estimation of a diffusive target in clustered wireless sensor networks	claim
376	192	Effective multi-resolution background subtraction	claim
377	192	On the relation between ICA and MMSE based source separation	claim
378	192	Non-linear tagging models with localist and distributed word representations	claim
379	192	Stochastic analysis of two-tier networks: Effect of spectrum allocation	claim
380	192	Optimal power allocation and joint source-channel coding for wireless DS-CDMA visual sensor networks using the Nash Bargaining Solution	claim
381	192	CROWDMOS: An approach for crowdsourcing mean opinion score studies	claim
382	191	Unsupervised determination of efficient Korean LVCSR units using a Bayesian Dirichlet process model	claim
383	191	Full-covariance UBM and heavy-tailed PLDA in i-vector speaker verification	claim
384	191	Convergence of a distributed parameter estimator for sensor networks with local averaging of the estimates	claim
385	191	Multipath model and exploitation in through-the-wall radar and urban sensing	claim
386	191	Efficient semidefinite relaxation for robust geolocation of unknown emitter by a satellite cluster using TDOA and FDOA measurements	claim
387	191	Online feature selection and classification	claim
388	191	Automatic audio tag classification via semi-supervised canonical density estimation	claim
389	190	A reversible jump MCMC algorithm for Bayesian curve fitting by using smooth transition regression models	claim
390	190	Language identification using a combined articulatory prosody framework	claim
391	190	Automatically finding semantically consistent n-grams to add new words in LVCSR systems	claim
392	190	An inner-product lower-bound estimate for dynamic time warping	claim
393	190	Adaptive frequency-domain biased estimation algorithms with automatic adjustment of shrinkage factors	claim
394	190	Utilizing glottal source pulse library for generating improved excitation signal for HMM-based speech synthesis	claim
395	190	Interference alignment in single-beam MIMO networks via homotopy continuation	claim
396	190	Distributed routing in networks using affinity propagation	claim
397	189	Detection and removal of binocular luster in compressed 3D images	claim
398	189	Degenerate Unmixing Estimation Technique using the Constant Q Transform	claim
399	189	A variable step size evolutionary affine projection algorithm	claim
400	189	Synthesis of ICA-based methods for localization of multiple broadband sound sources	claim
401	189	Accurate parameter generation using fixed-point arithmetic for embedded HMM-based speech synthesizers	claim
402	189	A flexible Speech Distortion Weighted Multi-channel Wiener Filter for noise reduction in hearing aids	claim
403	188	Time domain reconstruction of spatial sound fields using compressed sensing	claim
404	188	Online Kernel SVM for real-time fMRI brain state prediction	claim
405	188	On selecting the hyperparameters of the DPM models for the density estimation of observation errors	claim
406	188	Music algorithm to localize sources with unknown directivity in acoustic imaging	claim
407	188	Optimized capacity bounds for the MIMO relay channel	claim
408	188	Exploiting multipath for blind source separation with sensor arrays	claim
409	188	Estimation of the frequency dependent reverberation time by means of warped filter-banks	claim
410	187	Global variance modeling on frequency domain delta LSP for HMM-based speech synthesis	claim
411	187	Efficient distributed resampling for particle filters	claim
412	187	Empirical weighting for Blind Source Separation in a multiple-snapshots scenario	claim
413	187	Compressed classification of observation sets with linear subspace embeddings	claim
414	187	Reproduction of independent narrowband soundfields in a multizone surround system and its extension to speech signal sources	claim
415	187	Distributed auxiliary particle filters using selective gossip	claim
416	187	Time delay estimation in the time-frequency domain based on a line detection approach	claim
417	187	A robust clock synchronization algorithm for wireless sensor networks	claim
418	187	Identification of MISO nonlinear systems via the semiparametric approach	claim
419	186	On the improvement of image registration for high accuracy super-resolution	claim
420	186	Sentence simplification for spoken language understanding	claim
421	186	Distributed beamforming for multiuser peer-to-peer and multi-group multicasting relay networks	claim
422	186	Signal extrapolation using Empirical Mode Decomposition with financial applications	claim
423	186	Multichannel harmonic and percussive component separation by joint modeling of spatial and spectral continuity	claim
424	186	Construction of positive time-frequency distributions using dynamic copula	claim
425	186	An utterance comparison model for speaker clustering using factor analysis	claim
426	186	A basis method for robust estimation of constrained MLLR	claim
427	185	Maximum a posteriori based regularization parameter selection	claim
428	185	Error compensation and reliability based view synthesis	claim
429	185	A new metric for VQ-based speech enhancement and separation	claim
430	185	Analysis of the pilot contamination effect in very large multicell multiuser MIMO systems for physical channel models	claim
431	185	Binaural sound source separation motivated by auditory processing	claim
432	185	Improving kernel-energy trade-offs for machine learning in implantable and wearable biomedical applications	claim
433	185	Outlier-aware robust clustering	claim
434	185	Stationary Common Spatial Patterns: Towards robust classification of non-stationary EEG signals	claim
435	185	Efficient image reconstruction under sparsity constraints with application to MRI and bioluminescence tomography	claim
436	185	A space time array processing for passive geolocalization of radio transmitters	claim
437	185	Direction-resolved estimation of multipath parameters for UWB channels: A partially collapsed Gibbs sampler method	claim
438	184	A study of the effect of emotional state upon text-independent speaker identification	claim
439	184	Dual constrained TV-based regularization	claim
440	184	Acoustic model training for non-audible murmur recognition using transformed normal speech data	claim
441	184	Median filter with absolute value norm spatial regularization	claim
442	184	Nonbinary LDPC decoding by min-sum with Adaptive Message Control	claim
443	184	Basis Pursuit in sensor networks	claim
444	184	Multi-view and multi-objective semi-supervised learning for large vocabulary continuous speech recognition	claim
445	184	Multichannel EEG analysis based on multi-scale multi-information	claim
446	184	Emotion classification from speech using evaluator reliability-weighted combination of ranked lists	claim
447	183	Adaptive modelling with tunable RBF network using multi-innovation RLS algorithm assisted by swarm intelligence	claim
448	183	A model-based auditory scene analysis approach and its application to speech source localization	claim
449	183	Robust adaptive event detection in non-intrusive load monitoring for energy aware smart facilities	claim
450	183	A sparsity based criterion for solving the permutation ambiguity in convolutive blind source separation	claim
451	183	Joint transmitter and receiver design with adaptive beamforming in MIMO SC-FDMA systems	claim
452	183	How much feedback overhead is required for base station cooperative transmission to outperform non-cooperative transmission?	claim
453	183	NAP for high level language identification	claim
454	183	Robust underdetermined blind audio source separation of sparse signals in the time-frequency domain	claim
455	183	Concept-based classification for multi-document summarization	claim
456	183	Cooperative sensing with sequential ordered transmissions to secondary fusion center	claim
457	183	Distributed linear discriminant analysis	claim
458	183	Joint encoding of the waveform and speech recognition features using a transform codec	claim
459	182	Are all basis updates for lattice-reduction-aided MIMO detection necessary?	claim
460	182	Integrated Sidelobe Level of sets of Rotated Legendre Sequences	claim
461	182	Sequential Monte Carlo method for parameter estimation in diffusion models of affinity-based biosensors	claim
462	182	Quadratic optimization based small scale details extraction	claim
463	182	Resolving spatial sampling effects in parametric directional filtering	claim
464	182	Adjustable bandwidth filter design with generalized farrow structure	claim
465	182	On visually evoked potentials in eeg induced by multiple pseudorandom binary sequences for brain computer interface design	claim
466	182	Parallel implementation of Multi-dimensional Ensemble Empirical Mode Decomposition	claim
467	181	Deterministic compressed-sensing matrices: Where Toeplitz meets Golay	claim
468	181	Efficient NLMS and RLS algorithms for a class of nonlinear filters using periodic input sequences	claim
469	181	Sensing-aware classification with high-dimensional data	claim
470	180	Sparse coding of auditory features for machine hearing in interference	claim
471	180	An atlas-based deep brain structure segmentation method: from coarse positioning to fine shaping	claim
472	180	Reaching consensus in asynchronous WSNs: Algebraic approach	claim
473	180	Subspace-based direction finding using transmit energy focusing in MIMO radar with colocated antennas	claim
474	180	A combinatorial optimization framework for subset selection in distributed multiple-radar architectures	claim
475	180	Adaptive recursive FLANN filters for nonlinear active noise control	claim
476	180	Compressed learning of high-dimensional sparse functions	claim
477	180	Rapid joint speaker and noise compensation for robust speech recognition	claim
478	180	High accurate model-integration-based voice conversion using dynamic features and model structure optimization	claim
479	180	On the construction of low-pass filters on the unit sphere	claim
480	180	Multi-channel EEG compression based on matrix and tensor decompositions	claim
481	179	Multiantenna detection under noise uncertainty and primary user's spatial structure	claim
482	179	Log-gaussian cox processes of visual keypoints for sonar texture recognition	claim
483	179	Distributed blind source separation with an application to audio signals	claim
484	179	Semi-supervised handwritten digit recognition using very few labeled data	claim
485	179	Compressive sensing meets game theory	claim
486	179	A time-distributed phase space histogram for detecting transient signals	claim
487	179	Spectral magnitude minimum mean-square error binary masks for DFT based speech enhancement	claim
488	179	The MIT LL 2010 speaker recognition evaluation system: Scalable language-independent speaker recognition	claim
489	179	Horror video scene recognition via Multiple-Instance learning	claim
490	179	Performance bounds for tracking in a multipath environment	claim
491	178	A wavelet-based data imputation approach to spectrogram reconstruction for robust speech recognition	claim
492	178	Interference alignment in clustered ad hoc networks: High reliability regime and per-cluster aloha	claim
493	178	An interference-free representation of instantaneous frequency of periodic signals and its application to F0 extraction	claim
494	178	Blind extraction of improper quaternion sources	claim
495	178	Study of robustness of zero frequency resonator method for extraction of fundamental frequency	claim
496	178	A probabilistic pixel-based approach to detect humans in video streams	claim
497	178	On the sum rate of ZF detectors over correlated K fading MIMO channels	claim
498	177	Polyphonic music transcription using note onset and offset detection	claim
499	177	Integrating binaural cues and blind source separation method for separating reverberant speech mixtures	claim
500	177	Complex random matrices and multiple-antenna spectrum sensing	claim
501	177	Defining the controlling parameter in constrained discriminative linear transform for supervised speaker adaptation	claim
502	177	Factor graph-based structural equilibria in dynamical games	claim
503	177	Real-time and multi-view face tracking on mobile platform	claim
504	177	3D medical image coding with optimal channel protection for wireless transmission	claim
505	177	Logarithmic weak regret of non-Bayesian restless multi-armed bandit	claim
506	176	Audio watermarking for acoustic propagation in reverberant environments	claim
507	176	Content-aware TCP-friendly congestion control for multimedia transmission	claim
508	176	I-Divergence-based dereverberation method with auxiliary function approach	claim
509	176	Transmit and receive filters for MISO FBMC systems subjected to power constraints	claim
510	176	Closed-form parameterization of the Pareto boundary for the two-user MISO interference channel	claim
511	176	RSS-based sensor localization with unknown transmit power	claim
512	176	Optimal structure of memory models for lossless compression of binary image contours	claim
513	176	Analyzing the performance of error-prone relay networks	claim
514	176	Efficient transform coding of two-channel audio signals by means of complex-valued stereo prediction	claim
515	176	An unconstrained method for lip detection in color images	claim
516	176	Improved pos tagging for text-to-speech synthesis	claim
517	175	Joint frequency spectral lag representation for cross-frequency modulation analysis in the brain	claim
518	175	Deep belief nets for natural language call-routing	claim
519	175	Structured discriminative models for noise robust continuous speech recognition	claim
520	175	Generating compound words with high order n-gram information in large vocabulary speech recognition systems	claim
521	175	Flame detection method in video using covariance descriptors	claim
522	175	Discriminant binary data representation for speaker recognition	claim
523	175	Seam merging for image resizing with structure preservation	claim
524	175	Pronunciation variation modeling of non-native proper names by discriminative tree search	claim
525	175	Enhanced Poisson sum representation for alpha-stable processes	claim
526	175	Detection of anomalous events from unlabeled sensor data in smart building environments	claim
527	175	Prosodic control of unit-selection speech synthesis: A probabilistic approach	claim
528	175	Intra-session variability compensation and a hypothesis generation and selection strategy for speaker segmentation	claim
529	174	Video processing with scale-aware saliency: Application to Frame Rate Up-Conversion	claim
530	174	Analysis of non-coherent code tracking for NPSK systems in presence of noise and fading	claim
531	174	Rate distorsion analysis in a disparity compensated scheme	claim
532	174	Factor analysis based VTS and JUD noise estimation and compensation	claim
533	174	Sparse coding and dictionary learning based on the MDL principle	claim
534	174	Syllabification of conversational speech using Bidirectional Long-Short-Term Memory Neural Networks	claim
535	174	Fast damped gauss-newton algorithm for sparse and nonnegative tensor factorization	claim
536	174	How many known symbols are required for linear channel estimation in OFDM?	claim
537	174	Image noise-informative map for noise standard deviation estimation	claim
538	174	How efficient is estimation with missing data?	claim
539	174	A Radio Frequency Identification System for accurate indoor localization	claim
540	173	Human gait parameter estimation based on micro-doppler signatures using particle filters	claim
541	173	A proportionate adaptive algorithm with variable partitioned block length for acoustic echo cancellation	claim
542	173	Beamforming design for multi-user two-way relaying with MIMO amplify and forward relays	claim
543	173	Adaptive scalable layer filtering process for video scheduling over wireless networks based on MAC buffer management	claim
544	173	Whole word discriminative point process models	claim
545	173	Multi-rank processing for passive ranging in underwater acoustic environments subject to spatial coherence loss	claim
546	173	Intra-frame prediction with lapped transforms for image coding	claim
547	173	An evaluation of noise power spectral density estimation algorithms in adverse acoustic environments	claim
548	173	Recent progress in prosodic speaker verification	claim
549	173	Robust nonparametric regression by controlling sparsity	claim
550	173	Optimal SIR algorithm vs. fully adapted auxiliary particle filter: A matter of conditional independence	claim
551	173	Ordering for energy efficient estimation and optimization in sensor networks	claim
552	173	An unsupervised algorithm for hybrid/morphological signal decomposition	claim
553	173	Short and smooth sampling trajectories for compressed sensing	claim
554	172	Robust and low-cost cascaded non-linear acoustic echo cancellation	claim
555	172	Detection of elliptical particles in atomic force microscopy images	claim
556	172	Detecting human activities in retail surveillance using hierarchical finite state machine	claim
557	172	Fast network querying algorithm for searching large-scale biological networks	claim
558	172	Max-min fair rate control based on a saddle-point characterization of some perron roots	claim
559	172	Constrained discriminative mapping transforms for unsupervised speaker adaptation	claim
560	172	A single snapshot optimal filtering method for fundamental frequency estimation	claim
561	172	DSP evolution from a teaching point of view	claim
562	172	Frame-wise HMM adaptation using state-dependent reverberation estimates	claim
563	172	Rate-SSIM optimization for video coding	claim
564	172	Learning vocal tract variables with multi-task kernels	claim
565	171	MRF-based automatic image ordering and its application to mosaicing	claim
566	171	Computationally efficient regularized acoustic imaging	claim
567	170	Simplified EM channel estimation in LTE systems	claim
568	170	Risk management for trading in multiple frequencies	claim
569	169	Epileptic seizure prediction using the spatiotemporal correlation structure of intracranial EEG	claim
570	169	Application specific loss minimization using gradient boosting	claim
571	169	Statistical Resolution Limit for source localization in a MIMO context	claim
572	169	Graph-based sequential particle filtering in lossy networks: Single and multiple collaborative cameras	claim
573	169	An augmented Lagrangian method for video restoration	claim
574	169	Non-linear noise compensation for robust speech recognition using Gauss-Newton method	claim
575	169	Boosting video classification using cross-video signals	claim
576	169	Audio identification based on spectral modeling of bark-bands energy and synchronization through onset detection	claim
577	169	Novel hierarchical ALS algorithm for nonnegative tensor factorization	claim
578	169	Arccosine kernels: Acoustic modeling with infinite neural networks	claim
579	169	Occlusion-based depth ordering on monocular images with Binary Partition Tree	claim
580	169	Concurrent estimation of singing voice F0 and phonemes by using spectral envelopes estimated from polyphonic music	claim
581	169	Multi-sensor estimation and detection of phase-locked sinusoids	claim
582	169	A new video similarity measure model based on video time density function and dynamic programming	claim
583	168	Exposing duplicated regions affected by reflection, rotation and scaling	claim
584	168	Two effective and computationally efficient pure-pixel based algorithms for hyperspectral endmember extraction	claim
585	168	Anti-forensics for frame deletion/addition in MPEG video	claim
586	168	Hilbert envelope based features for robust speaker identification under reverberant mismatched conditions	claim
587	168	A Bayesian marked point process for object detection. Application to muse hyperspectral data	claim
588	168	Multi-graph regularization for efficient delivery of user generated content in online social networks	claim
589	168	Automatic minute generation for parliamentary speech using conditional random fields	claim
590	168	Reduced complexity blind deterministic frequency offset estimation in OFDM systems	claim
591	168	Low-rank matrix completion with geometric performance guarantees	claim
592	168	Named entity recognition from Conversational Telephone Speech leveraging Word Confusion Networks for training and recognition	claim
593	167	Drum extraction from polyphonic music based on a spectro-temporal model of percussive sounds	claim
594	167	Covariate-dependent dictionary learning and sparse coding	claim
595	167	Talker-to-listener distance effects on the variations of the intensity and the fundamental frequency of speech	claim
596	167	Discriminative training for Bayesian sensing hidden Markov models	claim
597	167	A unified approach to real time audio-to-score and audio-to-audio alignment using sequential Montecarlo inference techniques	claim
598	167	The compressive multiplexer for multi-channel compressive sensing	claim
599	167	Compression using self-similarity-based temporal super-resolution for full-exposure-time video	claim
600	167	Horizontal small target detection with cooperative background estimation and removal filters	claim
601	167	Identification of ARMA models using intermittent and quantized output observations	claim
602	167	Efficient search of music pitch contours using wavelet transforms and segmented dynamic time warping	claim
603	167	A perceptually transparent audio power reduction algorithm for loudspeaker power management	claim
604	167	Empirical divergence maximization for quantizer design: An analysis of approximation error	claim
605	166	Model-based speech enhancement using SNR dependent MMSE estimation	claim
606	166	Amplitude modulation spectrogram based features for robust speech recognition in noisy and reverberant environments	claim
607	166	Cooperation strategies for secrecy in MIMO relay networks with unknown eavesdropper CSI	claim
608	166	Using clustering comparison measures for speaker recognition	claim
609	166	Linear IIR-MMSE precoding for frequency selective MIMO channels	claim
610	166	Objective evaluation of the Dynamic Model Selection method for spectral voice conversion	claim
611	166	Improving head-related impulse response measured in noisy environments with spatio-temporal frequency analysis	claim
612	166	Multiple LDPC decoding using bitplane correlation for Transform Domain Wyner-Ziv video coding	claim
613	166	MPtracker: A new multi-pitch detection and separation algorithm for mixed speech signals	claim
614	166	Localization of non-linguistic events in spontaneous speech by Non-Negative Matrix Factorization and Long Short-Term Memory	claim
615	166	Fixed point iteration for max-min sir balancing with general interference functions	claim
616	166	A new structure with spectrum-tuning of residual noise for active noise control	claim
617	165	Robust frequency synchronization for an OFDMA uplink system disturbed by a Cognitive Radio system interference	claim
618	165	Dereverberation of ultrasound echo data in vascular imaging applications	claim
619	165	Revisiting adaptive least-squares estimation and application to online sparse signal recovery	claim
620	165	Single-carrier systems with MMSE linear equalizers: Performance degradation due to channel and CFO estimation errors	claim
621	165	Scalable robust hypothesis tests using graphical models	claim
622	165	Spatially consistent view synthesis with coordinate alignment	claim
623	165	Design of multipole loudspeaker array based on spherical harmonic expansion	claim
624	165	A sliding-window online fast variational sparse Bayesian learning algorithm	claim
625	165	Design of UEP-based MSE-minimizing rateless codes for source-channel coding	claim
626	165	Sub-Nyquist sampling of short pulses	claim
627	165	The SRI NIST 2010 speaker recognition evaluation system	claim
628	165	Widely distributed MIMO radar beamforming for detecting targets with slow RCS fluctuations	claim
629	165	Improved F0 modeling and generation in voice conversion	claim
630	165	FPGA implementation made easy for applied digital signal processing courses	claim
631	165	Simultaneous object tracking and depth estimation using color shifting property of a multiple color-filter aperture camera	claim
632	165	A modified MAP criterion based on hidden Markov model for voice activity detecion	claim
633	165	Non-parametric bayesian measurement noise density estimation in non-linear filtering	claim
634	165	Accelerating standards and measurements for the smart grid	claim
635	164	Stability analysis of multiplicative update algorithms for non-negative matrix factorization	claim
636	164	Joint source-filter modeling using flexible basis functions	claim
637	164	Sparse variable reduced rank regression via Stiefel optimization	claim
638	164	New radix-based FHT algorithm for computing the discrete Hartley transform	claim
639	164	Pitch transposition and breathiness modification using a glottal source model and its adapted vocal-tract filter	claim
640	164	MCMC inference of the shape and variability of time-response signals	claim
641	164	Decomposition of speech signals for analysis of aperiodic components of excitation	claim
642	164	On spatio-temporal Tomlinson Harashima Precoding in IIR channels: MMSE solution, properties, and fast computation	claim
643	164	Network-MIMO backhauling for QOS-constrained relay transmission	claim
644	164	A new approach for the estimation of the porosity in NMR	claim
645	164	Forensically inspired approaches to automatic speaker recognition	claim
646	164	SEAMLESS P2P-MDVC with well-balanced descriptions	claim
647	164	Texture removal by pixel classification using a rotating filter	claim
648	164	Polynomial expansion detector for uniform linear arrays	claim
649	164	Optimal and low-complexity iterative joint network/channel decoding for the multiple-access relay channel	claim
650	163	Recovery of sparse perturbations in Least Squares problems	claim
651	163	Classification by weighting for spatio-frequency components of EEG signal during motor imagery	claim
652	163	Asymptotically MMSE-optimum pilot design for comb-type OFDM channel estimation in high-mobility scenarios	claim
653	163	Classifying soundtracks with audio texture features	claim
654	163	Non-negative matrix deconvolution in noise robust speech recognition	claim
655	163	Reconstructing completely overlapped notes from musical mixtures	claim
656	163	Direct load management of electric vehicles	claim
657	163	Investigations into the incorporation of the Ideal Binary Mask in ASR	claim
658	163	Estimation of fundamental frequency from surface electromyographic data: EMG-to-F0	claim
659	163	Analysis-synthesis based speech enhancement with improved spectrum envelope estimation by tracking speech dynamics	claim
660	163	Robust speaker identification using a CASA front-end	claim
661	163	Robust parametrization for non-destructive evaluation of composites using ultrasonic signals	claim
662	163	Use of VTL-wise models in feature-mapping framework to achieve performance of multiple-background models in speaker verification	claim
663	163	On gradient type adaptive filters with non-symmetric matrix step-sizes	claim
664	163	Maximum marginal likelihood estimation for nonnegative dictionary learning	claim
665	163	Linear precoding for time-varying MIMO channels with low-complexity receivers	claim
666	162	Saturation-robust SAR image formation	claim
667	162	Distributed Gaussian particle filtering using likelihood consensus	claim
668	162	A methodology based on Transportation problem modeling for designing parallel interleaver architectures	claim
669	162	Accurate non-iterative depth layer extraction algorithm for image based rendering	claim
670	162	Enriching Mandarin speech recognition by incorporating a hierarchical prosody model	claim
671	162	Cooperative mobile network localization via subspace tracking	claim
672	162	Analysis of adaptive feedback and echo cancelation algorithms in a general multiple-microphone and single-loudspeaker system	claim
673	162	Estimating Sparse MIMO channels having Common Support	claim
674	162	Improved thresholds for rank minimization	claim
675	162	Nonnegative 3-way tensor factorization via conjugate gradient with globally optimal stepsize	claim
676	162	ON the recovery of time-varying spectral envelope information from AQHM-derived spectra	claim
677	162	Joint DOA and TDOA estimation for 3D localization of reflective surfaces using eigenbeam MVDR and spherical microphone arrays	claim
678	161	Adaptive N-normalization for enhancing music similarity	claim
679	161	Modeling nodal prices in deregulated electricity markets in the usa: current practices and future needs	claim
680	161	Performance evaluation of Raptor and Random Linear Codes for H.264/AVC video transmission over DVB-H networks	claim
681	161	Data-path and memory error compensation technique for low power JPEG implementation	claim
682	161	Compressed sensing signal recovery via A* Orthogonal Matching Pursuit	claim
683	161	Abnormal motion detection in crowded scenes using local spatio-temporal analysis	claim
684	160	Parametric modeling and linear estimation of elastic deformations	claim
685	160	Stochastic resource allocation for cognitive radio networks based on imperfect state information	claim
686	160	Incorporating alignments into Conditional Random Fields for grapheme to phoneme conversion	claim
687	160	Fourier expansion of hammerstein models for nonlinear acoustic system identification	claim
688	160	Detection of auditory stimulus onset in the Pontine Nucleus using a multichannel multi-unit activity electrode	claim
689	160	A general Bayesian algorithm for visual object tracking based on sparse features	claim
690	160	Cross-language bootstrapping based on completely unsupervised training using multilingual A-stabil	claim
691	160	Video error concealment using sparse recovery and local dictionaries	claim
692	159	Reconfigurable decoder architectures for Raptor codes	claim
693	159	Approximation of pattern transformation manifolds with parametric dictionaries	claim
694	159	Smoothing fir filtering of discrete state-space polynomial signal models	claim
695	159	Estimation of ordinal approach-avoidance labels in dyadic interactions: Ordinal logistic regression approach	claim
696	159	Sparse channel estimation with lp-norm and reweighted l1-norm penalized least mean squares	claim
697	159	Toward text message normalization: Modeling abbreviation generation	claim
698	159	Speaker authentication using video-based lip information	claim
699	158	A classifier-based decoding approach for large scale distributed coding	claim
700	158	Phase-sensitive speech enhancement for cochlear implant processing	claim
701	158	Speech enhancement using a joint map estimator with Gaussian mixture model for (non-)stationary noise	claim
702	158	Modified embedding for multi-regime detection in nonstationary streaming data	claim
703	158	Dart - a high level software-defined radio platform model for developing the run-time controller	claim
704	158	User admission in MIMO interference alignment networks	claim
705	158	An adaptive time-frequency resolution approach for Non-negative Matrix Factorization based single channel sound source separation	claim
706	158	Using latent topic features to improve binary classification of spoken documents	claim
707	157	An SVM based classification approach to speech separation	claim
708	157	Maximum likelihood SAR autofocus with low-return region	claim
709	157	On optimal channel training for uplink network MIMO systems	claim
710	157	Source-normalised-and-weighted LDA for robust speaker recognition using i-vectors	claim
711	157	Polar randomized hough transform for lane detection using loose constraints of parallel lines	claim
712	157	From direction of arrival estimates to localization of planar reflectors in a two dimensional geometry	claim
713	157	A study of an irrelevant variability normalization based discriminative training approach for LVCSR	claim
714	157	Continuous pilot based adaptive estimation for IDMA systems on underwater acoustic channels	claim
715	157	Worst-case based robust adaptive beamforming for general-rank signal models using positive semi-definite covariance constraint	claim
716	156	An optimization algorithm of independent mean and variance parameter tying structures for HMM-based speech synthesis	claim
717	156	Binaural extension and performance of single-channel spectral subtraction dereverberation algorithms	claim
718	156	Analog joint source-channel Multiple Description coding scheme over AWGN parallel channels	claim
719	156	Fundamental diversity, multiplexing, and array gain tradeoff under different MIMO channel models	claim
720	156	Gesture-based Dynamic Bayesian Network for noise robust speech recognition	claim
721	156	Low-complexity detection of golden codes in LDPC-coded OFDM systems	claim
722	156	Hybrid probabilistic adaptation mode controller for generalized sidelobe canceller-based target-directional speech enhancement	claim
723	156	Synthesis filter bank optimization with lattice structure constraints in 2D separable image processing	claim
724	156	Modeling temporal correlations in content fingerprints	claim
725	155	Automatic diagnosis of ADHD based on nonlinear analysis of actimetry registries	claim
726	155	Robust speech recognition using dynamic noise adaptation	claim
727	155	Directions-of-arrival estimation using a sparse spatial spectrum model with uncertainty	claim
728	155	Combined compressed sensing and parallel mri compared for uniform and random cartesian undersampling of K-space	claim
729	155	Design of Hilbert transform pairs of orthonormal wavelet bases with improved analyticity	claim
730	155	Mixture Kalman filtering for joint carrier recovery and channel estimation in time-selective Rayleigh fading channels	claim
731	155	A new method for visual stylometry on impressionist paintings	claim
732	155	Joint blind estimation of carrier phase and frequency offset for QAM signals using circular harmonic decomposition	claim
733	155	Reusing the H.264/AVC deblocking filter for efficient spatio-temporal prediction in video coding	claim
734	155	Image prediction based on non-negative matrix factorization	claim
735	155	Robust Bayesian Analysis applied to Wiener filtering of speech	claim
736	155	Fast speaker diarization based on binary keys	claim
737	155	Interference cancellation for OFDM systems with hierarchical modulation over non-linear satellite channels	claim
738	154	Automatic recognition of speech without any audio information	claim
739	154	Parallel Transformation Network features for speaker recognition	claim
740	154	Dynamics of tongue gestures extracted automatically from ultrasound	claim
741	154	Lattice-based unsupervised acoustic model training	claim
742	154	Language dependent universal phoneme posterior estimation for mixed language speech recognition	claim
743	154	Cross-spectrum and coherence function estimation using time-delayed Thomson multitapers	claim
744	154	Spatially-correlated sensor discriminant analysis	claim
745	154	Image compression using the Iteration-Tuned and Aligned Dictionary	claim
746	154	Global emergent behaviors in clouds of agents	claim
747	154	Adaptive detection of multiple point-like targets with conic acceptance	claim
748	153	Reinforcement learning for energy-efficient wireless transmission	claim
749	153	Learning non-parametric models of pronunciation	claim
750	153	Hierarchical audio classification using cepstral modulation ratio regressions based on Legendre polynomials	claim
751	153	Efficient discrete fractional Hirschman optimal transform and its application	claim
752	153	Dynamic spectrum management in DSL with asynchronous crosstalk	claim
753	153	A Lagrangian dual relaxation approach to ML MIMO detection: Reinterpreting regularized lattice decoding	claim
754	153	Decentralized support detection of multiple measurement vectors with joint sparsity	claim
755	152	Probabilistic SINR constrained robust transmit beamforming: A Bernstein-type inequality based conservative approach	claim
756	152	Topographic phase maps using iterative independent component analysis	claim
757	152	Tracking changes in continuous emotion states using body language and prosodic cues	claim
758	152	Spatio-temporal consistency in video disparity estimation	claim
759	152	Hybrid approach for multichannel source separation combining time-frequency mask with multi-channel Wiener filter	claim
760	152	Supervised nonlinear spectral unmixing using a polynomial post nonlinear model for hyperspectral imagery	claim
761	152	Extension of the Global Matched Filter to structured groups of atoms: Application to harmonic signals	claim
762	152	One sentence voice adaptation using GMM-based frequency-warping and shift with a sub-band basis spectrum model	claim
763	152	System-theoretic formulation and analysis of dynamic consensus propagation	claim
764	152	A simplified Subspace Gaussian Mixture to compact acoustic models for speech recognition	claim
765	152	Target tracking and localization with ambiguous phase measurements of sensor networks	claim
766	152	Powerful extensions to CRFS for grapheme to phoneme conversion	claim
767	152	Exemplar-based Sparse Representation phone identification features	claim
768	152	Using residual vector quantization for image content classification	claim
769	151	A new error resilience scheme based on FMO and error concealment in H.264/AVC	claim
770	151	Soft frame margin estimation of Gaussian Mixture Models for speaker recognition with sparse training data	claim
771	151	Time-frequency reassigned features for automatic chord recognition	claim
772	151	Iterative estimation of structures of multiple RNA homologs: Turbofold	claim
773	151	Near-field array shape calibration	claim
774	151	Generalizations of Blom And Bloem's PDF decomposition for permutation-invariant estimation	claim
775	151	Simultaneous dialog act segmentation and classification from human-human spoken conversations	claim
776	150	Snapshot spectral imaging via compressive random convolution	claim
777	150	A reweighted least squares algorithm for detection of QAM signaling in MIMO channels	claim
778	150	An optimal filtering for unmasked noise prevention	claim
779	150	Weighted compressed sensing and rank minimization	claim
780	150	Subspace pursuit method for kernel-log-linear models	claim
781	150	Performance optimized predictor blending technique for lossless image coding	claim
782	150	Simultaneous processing of sound source separation and musical instrument identification using Bayesian spectral modeling	claim
783	150	Joint modeling of observed inter-arrival times and waveform data with multiple hidden states for neural spike-sorting	claim
784	150	A flexible high-throughput hardware architecture for a gaussian noise generator	claim
785	149	Structural similarity indices for high dynamic range imaging	claim
786	149	Voxel-based Viterbi Active Speaker Tracking (V-VAST) with best view selection for video lecture post-production	claim
787	149	Speaker diarization of meetings based on speaker role n-gram models	claim
788	149	Gaussian approximation of the LLR distribution for the ML and partial marginalization MIMO detectors	claim
789	149	Parameter estimation using sparse reconstruction with dynamic dictionaries	claim
790	149	Diffusions of innovations on deterministic topologies	claim
791	149	An analytical approach to local sound field synthesis using linear arrays of loudspeakers	claim
792	148	Audio recognition in the wild: Static and dynamic classification on a real-world database of animal vocalizations	claim
793	148	Analytical performance assessment of 1-D Structured Least Squares	claim
794	148	Improved spoken term detection using support vector machines based on lattice context consistency	claim
795	148	Improved spoken term detection with graph-based re-ranking in feature space	claim
796	148	A knowledge-based algorithm to remove blocking artifacts in skin images for forensic analysis	claim
797	148	Image inpainting-based edge enhancement using the eikonal equation	claim
798	148	Evaluating music sequence models through missing data	claim
799	148	Improved estimation of the amplitude envelope of time-domain signals using true envelope cepstral smoothing	claim
800	148	A comparative analysis of biometric secret-key binding schemes based on QIM and Wyner-Ziv coding	claim
801	148	Perceiving graphical and pictorial information via touch and hearing	claim
802	148	Speech enhancement with masking properties in eigen-domain for colored noise	claim
803	148	Labeling skin tissues in ultrasound images using a generalized Rayleigh mixture model	claim
804	148	Speaker similarity evaluation of foreign-accented speech synthesis using HMM-based speaker adaptation	claim
805	148	Driver risk evaluation based on acceleration, deceleration, and steering behavior	claim
806	147	A design procedure for oversampled nonuniform filter banks with perfect-reconstruction	claim
807	147	Acoustic-to-articulatory inversion using an episodic memory	claim
808	147	Monitoring workspace activities using accelerometers	claim
809	147	Region-adaptive probability model selection for the arithmetic coding of video texture	claim
810	147	USPACOR: Universal sparsity-controlling outlier rejection	claim
811	147	A supervised approach to movie emotion tracking	claim
812	147	Comparison of several covariance matrix estimators for portfolio optimization	claim
813	147	Using stacked transformations for recognizing foreign accented speech	claim
814	147	Sequential Monte Carlo Radio-Frequency tomographic tracking	claim
815	147	Acoustic-phonetic information from excitation source for refining manner hypotheses of a phone recognizer	claim
816	147	Temporal hysteresis model of time varying subjective video quality	claim
817	147	Improved speaker recognition when using i-vectors from multiple speech sources	claim
818	147	Sparsity-based Sinogram Denoising for low-dose Computed Tomography	claim
819	147	Time-to-Onset latency in fMRI: Fast detection of delayed activation	claim
820	147	Generalized interior-point method for constrained peak power minimization of OFDM signals	claim
821	147	A wideband doubly-sparse approach for MITO sparse filter estimation	claim
822	146	Speech inversion: Benefits of tract variables over pellet trajectories	claim
823	146	A sparse covariance-based method for direction of arrival estimation	claim
824	146	Efficient iterative receiver for bit-Interleaved Coded Modulation according to the DVB-T2 standard	claim
825	146	A metric approach toward point process divergence	claim
826	146	Joint blind source separation from second-order statistics: Necessary and sufficient identifiability conditions	claim
827	146	Closed-form approximation of linear phase IIR digital filters with guaranteed stability	claim
828	146	Cooperative sensing in cognitive networks under malicious attack	claim
829	146	Kernel cross-modal factor analysis for multimodal information fusion	claim
830	146	Spiral colour model: Reduction from 3-D to 2-D	claim
831	146	Generalized low-rank decompositions with switching and adaptive algorithms for space-time adaptive processing	claim
832	146	New results on adaptive computational resource allocation in soft MIMO detection	claim
833	145	Particle algorithms for filtering in high dimensional state spaces: A case study in group object tracking
834	145	Dynamic signal combining for distributed microphone systems in car environments	claim
835	145	Stochastic transceiver design in multi-antenna channels with statistical channel state information	claim
836	145	Detection of targets embedded in multipath clutter with Time Reversal	claim
837	145	Multi-microphone interference suppression using the principal subspace modification and its application to speech recognition	claim
838	145	Jump Function Kolmogorov for overlapping audio event classification	claim
839	145	Variational methods for spectral unmixing of hyperspectral images	claim
840	145	Joint disparity and optical flow by correspondence growing	claim
841	145	A learning-based approach to explosives detection using Multi-Energy X-Ray Computed Tomography	claim
842	145	A novel algorithm of seeking FrFT order for speech processing	claim
843	145	Cooperative Maximum Likelihood estimation for fluid flow dynamics in biosensor arrays	claim
844	145	On the design of matched orthonormal wavelets with compact support	claim
845	144	Non-parallel training for voice conversion based on FT-GMM	claim
846	144	Local probability distribution of natural signals in sparse domains	claim
847	144	Graphical models for localization of the seizure focus from interictal intracranial EEG	claim
848	144	Robust GNSS signal detection in the presence of navigation data bits	claim
849	144	Parallel computation of adaptive lattice filters	claim
850	144	Cross-Channel Spectral Subtraction for meeting speech recognition	claim
851	144	Multi-rate polyphase DSP and LMS calibration schemes for oversampled data conversion systems	claim
852	144	Practical codes for lossy compression when side information may be absent	claim
853	144	Exploiting sparseness of backing-off language models for efficient look-ahead in LVCSR	claim
854	144	Social norm based incentive mechanisms for peer-to-peer networks	claim
855	144	Machine and acoustical condition dependency analyses for fast acoustic likelihood calculation techniques	claim
856	143	Automatic musical thumbnailing based on audio object localization and its evaluation	claim
857	143	Speech bandwidth extension using Gaussian mixture model-based estimation of the highband mel spectrum	claim
858	143	Enhanced coding of high-frequency tonal components in MPEG-D USAC through joint application of ESBR and sinusoidal modeling	claim
859	143	Estimation of cortical connectivity from E/MEG using nonlinear state-space models	claim
860	143	Transmit beamspace design for direction finding in colocated MIMO radar with arbitrary receive array	claim
861	143	Non-data-aided adaptive beamforming algorithm based on the Widely Linear Auxiliary Vector Filter	claim
862	143	Improved transient oscillation detection with multiwavelets	claim
863	143	Language-independent constrained cepstral features for speaker recognition	claim
864	143	Exploiting query click logs for utterance domain detection in spoken language understanding	claim
865	143	Score informed audio source separation using a parametric model of non-negative spectrogram	claim
866	143	Causal signal recovery from U-invariant samples	claim
867	143	Factored covariance modeling for text-independent speaker verification	claim
868	143	Time prediction of non flat fading channels	claim
869	143	A new speaker identification algorithm for gaming scenarios	claim
870	143	Online performance guarantees for sparse recovery	claim
871	142	Multilayer perceptron with sparse hidden outputs for phoneme recognition	claim
872	142	A fast solution to robust minimum variance beamformer and application to simultaneous MEG and local field potential	claim
873	142	Impulsive interference mitigation in ad hoc networks based on alpha-stable modeling and particle filtering	claim
874	142	Maximum likelihood adaptation of histogram equalization with constraint for robust speech recognition	claim
875	142	Finding curves in SAR CCD images	claim
876	142	Multivariate texture retrieval using the SIRV representation and the geodesic distance	claim
877	142	Delayless soft-decision decoding of high-quality audio transmitted over awgn channels	claim
878	142	The asymptotic properties of polynomial phase estimation by least squares phase unwrapping	claim
879	142	Linear time decoding of real-field codes over high error rate channels	claim
880	142	Scheduling of CAL actor networks based on dynamic code analysis	claim
881	142	Increasing discriminative capability on MAP-based mapping function estimation for acoustic model adaptation	claim
882	142	Polyphonic audio-to-score alignment based on Bayesian Latent Harmonic Allocation Hidden Markov Model	claim
883	141	A comparative analysis of dynamic network decoding	claim
884	141	Evolutive method based on a generalized eigenvalue decomposition to estimate time varying autoregressive parameters from noisy observations	claim
885	141	Towards robust word discovery by self-similarity matrix comparison	claim
886	141	A sampling-based environment population projection approach for rapid acoustic model adaptation	claim
887	141	Multi-class Model M	claim
888	141	An analysis of machine translation and speech synthesis in speech-to-speech translation system	claim
889	141	Transceiver optimization for multi-user multi-antenna two-way relay channels	claim
890	141	A low-power implantable neuroprocessor on nano-FPGA for Brain Machine interface applications	claim
891	141	Polytope kernel density estimates on Delaunay graphs	claim
892	141	Semi-algebraic canonical decomposition of multi-way arrays and Joint Eigenvalue Decomposition	claim
893	140	A time-warping framework for speech turbulence-noise component estimation during aperiodic phonation	claim
894	140	Direction-of-arrival estimation and array calibration for partly-calibrated arrays	claim
895	140	Progress in example based automatic speech recognition	claim
896	140	Bio-inspired cooperative optimization with application to bacteria motility	claim
897	140	Convex relaxation approaches to maximum likelihood DOA estimation in ULA's and UCA's with unknown mutual coupling	claim
898	140	Design and analysis of a narrowband filter for optical platform	claim
899	140	Time-varying lung ventilation analysis of 4DCT using image registration	claim
900	140	Asymptotic performance of distributed detection over random networks	claim
901	140	Location-aware query parsing for mobile voice search	claim
902	140	Correlation properties of quantization noise	claim
903	140	A methodology for evaluating the accuracy of wave field rendering techniques	claim
904	140	Data driven model based least squares image reconstruction for radio astronomy	claim
905	140	On the implementation of MIMO-OFDM schemes using perturbation of the QR decomposition: Application to 3GPP LTE-a systems	claim
906	140	A simple music/voice separation method based on the extraction of the repeating musical structure	claim
907	140	Speech processing with a cortical representation of audio	claim
908	140	Incremental-cooperative strategies in combination of adaptive filters	claim
909	139	Learning sparse dictionaries with a popularity-based model	claim
910	139	Acoustic vector-sensor beamforming in the presence of flow noise	claim
911	139	Combining HMM-based melody extraction and NMF-based soft masking for separating voice and accompaniment from monaural audio	claim
912	139	SVM feature selection for multidimensional EEG data	claim
913	139	Collision resolution in multiple access networks with physical-layer network coding and distributed fountain coding	claim
914	139	Discrete regression methods on the cone of positive-definite matrices	claim
915	139	Promoting convergence in multi-channel blind signal separation using PNLMS	claim
916	139	Perceptual differentiation modeling explains phoneme mispronunciation by non-native speakers	claim
917	139	Well-calibrated heavy tailed Bayesian speaker verification for microphone speech	claim
918	139	Similarity learning for semi-supervised multi-class boosting	claim
919	139	Pilot optimization for time-delay and channel estimation in OFDM systems	claim
920	138	Real-time design of a space/spatial-frequency optimal filter for highly nonstationary two-dimensional signal estimation	claim
921	138	Natural gradient approach in orthogonal matrix optimization using cayley transform	claim
922	138	Eigentriphones: A basis for context-dependent acoustic modeling	claim
923	138	Wavelet footprints for detection and sorting of extracellular neural action potentials	claim
924	138	Iterative hard thresholding for compressed sensing with partially known support	claim
925	138	Informative dialect recognition using context-dependent pronunciation modeling	claim
926	138	A hierarchical generative model for Generic Audio Document Categorization	claim
927	138	Versatile and portable DSP platform for learning embedded signal processing	claim
928	138	Preserve ordering property of generated LSPS for minimum generation error training in HMM-based speech synthesis	claim
929	138	Space-time beamforming for multiuser wireless relay networks	claim
930	138	An iterative strategy to approach corners using a new saliency measurement	claim
931	138	An approximate Minimum MOSPA estimator	claim
932	137	Time-constrained sequential pattern discovery for music genre classification	claim
933	137	High-throughput implementation of tree-search algorithms for vector precoding	claim
934	137	Resolving FD-BSS permutation for arbitrary array in presence of spatial aliasing	claim
935	137	Linear Predictive perceptual filtering for Acoustic Vector Sensors: Exploiting directional recordings for high quality speech enhancement	claim
936	137	Training design in single relay AF cooperative systems with correlated channels	claim
937	137	A near-optimal least squares solution to received signal strength difference based geolocation	claim
938	137	Anchor selection with anchor location uncertainty in wireless sensor network localization	claim
939	137	Continuous audio analytics by HMM and Viterbi decoding	claim
940	137	Design of robust steerable broadband beamformers incorporating microphone gain and phase error characteristics	claim
941	137	Survey and evaluation of acoustic features for speaker recognition	claim
942	136	A logarithmic based pole-zero vocal tract model estimation for speaker verification	claim
943	136	Error exponents for decentralized detection in feedback architectures	claim
944	136	Sparse video recovery using Linearly Constrained Gradient Projection	claim
945	136	On the effect of amr and AMR-WB GSM compression on overlapped speech for forensic analysis	claim
946	136	SRF: Matrix completion based on smoothed rank function	claim
947	135	Identification and compensation of Wiener-Hammerstein systems with feedback	claim
948	135	Fast algorithm for beamforming problems in distributed communication of relay networks	claim
949	135	User verification: Matching the uploaders of videos across accounts	claim
950	135	Using multiple visual tandem streams in audio-visual speech recognition	claim
951	135	Non-stationary noise estimation method based on bias-residual component decomposition for robust speech recognition	claim
952	135	Blind separation of multiple binary sources from one nonlinear mixture	claim
953	135	The transformed Variational Bayes approximation	claim
954	134	Message-passing for base-calling in sequencing-by-synthesis systems	claim
955	134	Source number estimation in impulsive noise environments using bootstrap techniques and robust statistics	claim
956	134	On the success of network inference using a markov routing model	claim
957	134	A pitch based noise estimation technique for robust speech recognition with Missing Data	claim
958	134	Speaker verification by inexperienced and experienced listeners vs. speaker verification system	claim
959	134	Error-entropy based channel state estimation of spatially correlated MIMO-OFDM	claim
960	134	Explicit recursivity into reproducing kernel Hilbert spaces	claim
961	134	Modeling anatomical heterogeneity in populations	claim
962	134	Linear manifold approximation based on differences of tangents	claim
963	134	Guaranteed error correction based on Fourier Compressive Sensing and Projective Geometry	claim
964	134	A system approach to residual echo suppression in robust hands-free teleconferencing	claim
965	134	Stereo audio classification for audio enhancement	claim
966	133	Towards improving network flow watermarks using the repeat-accumulate codes	claim
967	133	Time-based localization for asynchronous wireless sensor networks	claim
968	133	Distributed multiaccess in hierarchical cognitive radio networks	claim
969	133	Signal reconstruction from sine wave crossings	claim
970	133	Glottal inverse filtering using stabilised weighted linear prediction	claim
971	132	Lorentzian based iterative hard thresholding for compressed sensing	claim
972	132	A conditional model for triggering understanding actions in a speech understanding system	claim
973	132	A robust quantization method using a robust Chinese remainder theorem for secret key generation	claim
974	132	Training of error-corrective model for ASR without using audio data	claim
975	132	Phoneme recognition using Boosted Binary Features	claim
976	132	Robust speech recognition using multiple prior models for speech reconstruction	claim
977	132	Seam carving with rate-dependent seam path information	claim
978	132	Language identification for singing	claim
979	131	An innovative RTP-based residential power scheduling scheme for smart grids	claim
980	131	Sentence level emotion recognition based on decisions from subsentence segments	claim
981	131	Direction-of-arrival estimation using acoustic vector sensors in the presence of noise	claim
982	131	On single-channel noise reduction in the time domain	claim
983	131	Detection of synthetic speech for the problem of imposture	claim
984	131	Group testing meets traitor tracing	claim
985	131	Equalization of multichannel acoustic system using sub-systems for speech dereverberation	claim
986	131	Sampling on locally defined principal manifolds	claim
987	131	Multidimensional ICA and its performance analysis applied to CMB observations	claim
988	130	Robust localization of multiple sources in reverberant environments using EB-ESPRIT with spherical microphone arrays	claim
989	130	Log-spectral amplitude estimation with Generalized Gamma distributions for speech enhancement	claim
990	130	Speaker characterization using spectral subband energy ratio based on Harmonic plus Noise Model	claim
991	130	Distributed training of large scale exponential language models	claim
992	130	Generating avatar's facial expressions from emotional states in daily conversation	claim
993	130	Computer-assisted transcription of speech based on confusion network reordering	claim
994	130	An evolutionary game-theoretic approach for image interpolation	claim
995	130	Fast identification of JPEG 2000 images for digital cinema profiles	claim
996	130	Efficient implementation of probabilistic multi-pitch tracking	claim
997	130	Recursive estimation of room impulse responses with energy conservation constraints	claim
998	130	Noise-robust F0 estimation using SNR-weighted summary correlograms from multi-band comb filters	claim
999	130	Random block-angular matrices for distributed data storage	claim
1000	130	Prediction of discrete cosine transformed coefficients in resized pixel blocks	claim
1001	129	Multistream speaker diarization through Information Bottleneck system outputs combination	claim
1002	129	Improved DCT coefficient analysis for forgery localization in JPEG images	claim
1003	129	POMDP concept policies and task structures for hybrid dialog management	claim
1004	129	Dispersion measures and entropy for seizure detection	claim
1005	129	Variational approximation of long-span language models for lvcsr	claim
1006	129	Improved Secondary Range Compression focusing method in GEO SAR	claim
1007	129	Mixed norms with overlapping groups as signal priors	claim
1008	129	Perfect Root-Of-Unity Codes with prime-size alphabet	claim
1009	128	Separating sources from sequentially acquired mixtures of heart signals	claim
1010	128	SIC receiver for polyphonic piano music	claim
1011	128	Dual frame of frequency warping operators	claim
1012	128	Proportionate-type normalized least mean square algorithm with gain allocation motivated by minimization of mean-square-weight deviation for colored input	claim
1013	128	Evaluation of objective measures for intelligibility prediction of HMM-based synthetic speech in noise	claim
1014	128	A novel approach using modulation features for multiphone-based speech recognition	claim
1015	128	Microphone position optimization for planar superdirective beamforming	claim
1016	128	HMM-based speech synthesiser using the LF-model of the glottal source	claim
1017	128	Nonstationary and temporally correlated source separation using Gaussian process	claim
1018	127	Continuous F0 in the source-excitation generation for HMM-based TTS: Do we need voiced/unvoiced classification?	claim
1019	127	Quantization and power allocation in wireless sensor networks with correlated data	claim
1020	127	Estimation of symmetric chi-square divergence for point processes	claim
1021	127	Effect of fast AGC on cochlear implant speech intelligibility	claim
1022	127	Impact of the correlation between forward and backscatter channels on RFID system performance	claim
1023	127	Extracting call-reason segments from contact center dialogs by using automatically acquired boundary expressions	claim
1024	127	Urban terrain tracking in high clutter with waveform-agility	claim
1025	127	Linearization ability evaluation of nonlinear filters employing dynamic distortion measurement	claim
1026	126	Spectral-envelope and group-delay models for transient signals - Applications to castanets and stop consonants	claim
1027	126	Time-frequency analysis compensating missing data for Atrial Fibrillation ECG assessment	claim
1028	125	Generalized geometric mean decomposition and DFE MMSE transceiver design for cyclic prefix systems	claim
1029	125	Soundtrack classification by transient events	claim
1030	125	Bayesian detection of interference in satellite navigation systems	claim
1031	125	Gear scale estimation for synthetic speed pulse generation	claim
1032	125	On cooperative image denoising	claim
1033	125	High-order center-frequency adaptive filters using block-diagram-based frequency transformation	claim
1034	124	Vocal attractiveness of statistical speech synthesisers	claim
1035	124	Global convergence of independent component analysis based on semidefinite programming relaxation	claim
1036	124	A new variational method for preserving point-like and curve-like singularities in 2-D images	claim
1037	124	A general proof of convergence for adaptive distributed beamforming schemes	claim
1038	124	On the instantaneous frequency smoothing for signals with quasi-linear frequency changes	claim
1039	124	Hidden Discrete Tempo Model: A tempo-aware timing model for audio-to-score alignment	claim
1040	124	Hardware acceleration of iterative image reconstruction for X-ray computed tomography	claim
1041	124	Bilingual audio-subtitle extraction using automatic segmentation of movie audio	claim
1042	123	A kernelized maximal-figure-of-merit learning approach based on subspace distance minimization	claim
1043	123	Lexical access experiments with context-dependent articulatory feature-based models	claim
1044	123	Handling verbose queries for spoken document retrieval	claim
1045	123	Maximum likelihood ICA of quaternion Gaussian vectors	claim
1046	123	Performance analysis of optimal beamforming in MIMO dual-hop amplify-and-forward systems	claim
1047	123	Regularized split gradient method for nonnegative matrix factorization	claim
1048	122	Adaptive reconstruction method of missing textures based on perceptually optimized algorithm	claim
1049	122	An efficient variable step-size proportionate affine projection algorithm	claim
1050	122	Grassmannian predictive coding for limited feedback multiuser MIMO systems	claim
1051	122	Hill climbing on speech lattices: A new rescoring framework	claim
1052	122	Improving text-independent phonetic segmentation based on the Microcanonical Multiscale Formalism	claim
1053	122	Efficient convex optimization for real-time robust beamforming with microphone arrays	claim
1054	121	A general framework for robust HOSVD-based indexing and retrieval with high-order tensor data	claim
1055	121	Sum-rate maximization of two-way amplify-and-forward relay networks with imperfect channel state information	claim
1056	121	Further analysis of latent affective mapping for naturally expressive speech synthesis	claim
1057	121	Nonlinear audio recurrence analysis with application to genre classification	claim
1058	121	Experiments in context-independent recognition of non-lexical 'yes' or 'no' responses	claim
1059	121	Gain and phase autocalibration for uniform rectangular arrays	claim
1060	121	Allophonic variations in visual speech synthesis for corrective feedback in CAPT	claim
1061	121	Interpolation based on stationary and adaptive AR(1) modeling	claim
1062	121	Stretching, compression and shearing disparity compensated prediction techniques for stereo and multiview video coding	claim
1063	121	A mode preserving particle filter	claim
1064	121	Practical limits in RSS-based positioning	claim
1065	120	Variational Bayesian Kalman filtering in dynamical tomography	claim
1066	120	Fast discriminative speaker verification in the i-vector space	claim
1067	120	An efficient rank-deficient computation of the Principle of Relevant Information	claim
1068	120	Channel quantization design in multiuser MIMO systems: Asymptotic versus practical conclusions	claim
1069	120	Semantic data selection for vertical business voice search	claim
1070	120	A novel decision function and the associated decision-feedback learning for speech translation	claim
1071	120	A novel analytical approach to orthogonal bases extraction from AM-FM signals	claim
1072	119	A multi-stream ASR framework for BLSTM modeling of conversational speech	claim
1073	119	Non-centralized target tracking with mobile agents	claim
1074	119	Zero-forcing and MMSE filters design on the 2-sphere	claim
1075	119	Stochastic optimization based on the Laplace transform order with applications to precoder designs	claim
1076	119	A simple network-power-saving resource allocation method for OFDMA cellular networks with multiple relays	claim
1077	119	Hardware architectures for successive cancellation decoding of polar codes	claim
1078	119	Bilingual acoustic modeling with state mapping and three-stage adaptation for transcribing unbalanced code-mixed lectures	claim
1079	119	Robust talking face video verification using joint factor analysis and sparse representation on GMM mean shifted supervectors	claim
1080	118	Feature normalization for speaker verification in room reverberation	claim
1081	118	Multicast transmit beamforming using a randomize-in-time strategy	claim
1082	118	Sparse frequency waveform design based on PSD fitting	claim
1083	118	The Rotational Lasso	claim
1084	118	Distributed beamforming for OFDM-based cooperative relay networks under total and per-relay power constraints	claim
1085	118	Frequency selective pitch transposition of audio signals	claim
1086	118	Quantifying perturbations in temporal dynamics for automated assessment of spastic dysarthric speech intelligibility	claim
1087	118	USSS-MITLL 2010 human assisted speaker recognition	claim
1088	118	Eigenspace sparsity for compression and denoising	claim
1089	118	Blind periodically time-varying MMOE channel shortening for OFDM systems	claim
1090	117	Online speech source separation based on maximum likelihood of local Gaussian modeling	claim
1091	117	Audio segmentation of broadcast news: A hierarchical system with feature selection for the Albayzin-2010 evaluation	claim
1092	117	Classifier subset selection and fusion for speaker verification	claim
1093	117	Pronunciation variants generation using SMT-inspired approaches	claim
1094	117	Wavelet domain detection of rust in steel bridge images	claim
1095	116	Know Thy Neighbor: Combining audio features and social tags for effective music similarity	claim
1096	116	A new criterion for optimal constrained minimax detection and classification	claim
1097	116	Speaker and noise factorisation on the AURORA4 task	claim
1098	116	Blind channel estimation for MPSK-based amplify-and-forward two-way relaying	claim
1099	116	A binaural algorithm for space and pitch detection	claim
1100	116	Pricing game and evolution dynamics for mobile video streaming	claim
1101	115	A data-driven residual gain approach for two-stage speech enhancement	claim
1102	115	Optimal radio access in femtocell networks based on markov modeling of interferers' activity	claim
1103	115	Performance analysis of a randomly spaced wireless microphone array	claim
1104	115	A FPGA architecture for real-time processing of variable-length FFTS	claim
1105	115	Speaker diarization of heterogeneous web video files: A preliminary study	claim
1106	115	Detection of upper airway narrowing via classification of LPC coefficients: Implications for obstructive sleep apnea diagnosis	claim
1107	115	Polyphase filters - A model for teaching the art of discovery in DSP	claim
1108	115	Bias analysis of the forced choice detection test for Rayleigh/Rician statistics	claim
1109	114	Performance analysis of MDL criterion for the detection of noncircular or/and nonGaussian components	claim
1110	114	Robust representations of cortical speech and language information	claim
1111	114	A constrained optimization approach for multi-zone surround sound	claim
1112	114	Tone detection of non-uniformly undersampled signals with frequency excision	claim
1113	114	Posterior features for template-based ASR	claim
1114	114	Noise power gain as a measure of errors in discrete-time transversal estimators	claim
1115	114	On efficient soft-input soft-output encoding of convolutional codes	claim
1116	114	Vocalistener2: A singing synthesis system able to mimic a user's singing in terms of voice timbre changes as well as pitch and dynamics	claim
1117	114	Geometric programming for aggregation of binary classifiers	claim
1118	114	Online learning with minority class resampling	claim
1119	114	A paired test for recognizer selection with untranscribed data	claim
1120	114	From maximum likelihood to iterative decoding	claim
1121	114	Signal recovery in shift-invariant spaces from partial frequency data	claim
1122	114	Iterative feature normalization for emotional speech detection	claim
1123	113	BEP walls for collaborative spectrum sensing	claim
1124	113	Support vector regression fusion scheme in phone duration modeling	claim
1125	113	Audio-visual synchronization recovery in multimedia content	claim
1126	113	A fast projected gradient optimization method for real-time perception-based clipping of audio signals	claim
1127	113	Converting 5.1 audio recordings to B-format for directional audio coding reproduction	claim
1128	112	A nativeness classifier for TED Talks	claim
1129	112	Detecting oscillating singularities in multifractal analysis: Application to hydrodynamic turbulence	claim
1130	112	On optimal precoding in wireless multicast systems	claim
1131	112	Offending frequency suppression with a reset algorithm to improve feedback cancellation in digital hearing aids	claim
1132	112	Exploring implicit score normalization techniques in speaker verification	claim
1133	112	Separation and tracking of multiple speakers in a reverberant environment using a multiple model particle filter glimpsing method	claim
1134	112	Mixed-integer linear programming framework for max-min power control with single-stage interference cancellation	claim
1135	112	Switching linear dynamic transducer for stereo data based speech feature mapping	claim
1136	112	Data-driven fMRI group classification using connected components and Gaussian process classifiers	claim
1137	111	Prior knowledge-based direction of arrival estimation	claim
1138	111	Multi-source TDOA estimation using SNR-based angular spectra	claim
1139	111	Fully non-local super-resolution via spectral hashing	claim
1140	111	Beating nyquist through correlations: A constrained random demodulator for sampling of sparse bandlimited signals	claim
1141	111	Robust speaker turn role labeling of TV Broadcast News shows	claim
1142	111	Entropy-constrained quantization of exponentially damped sinusoids parameters	claim
1143	111	Using F0 to constrain the unit selection Viterbi network	claim
1144	111	Real-time software implementation of an IEEE 802.11a baseband receiver on Intel multicore	claim
1145	111	Joint data detection and dominant singular mode estimation in time varying reciprocal MIMO systems	claim
1146	111	Significance of vowel epenthesis in Telugu text-to-speech synthesis	claim
1147	111	Rapid feature space MLLR speaker adaptation with bilinear models	claim
1148	111	Phase-based information for voice pathology detection	claim
1149	110	Quantizer design with mismatched side information at the decoder	claim
1150	110	OFO estimation methods with wide acquisition ranges for MB-OFDM-based UWB systems	claim
1151	110	Online anomaly detection with expert system feedback in social networks	claim
1152	109	Power allocation for orthogonal AF relay systems with outage-based QOS constraints	claim
1153	109	Transitional surrogates	claim
1154	109	An adaptive noise canceller with adaptive delay compensation for a distant noise source	claim
1155	109	Improved models for Mandarin speech-to-text transcription	claim
1156	109	Quantization with an adjustable codeword length penalty	claim
1157	109	Very low bit-rate F0 coding for phonetic vocoder using MSD-HMM with quantized F0 context	claim
1158	109	A family of algorithms for blind equalization of QAM signals	claim
1159	109	Class of double-talk detectors based on the holder inequality	claim
1160	109	Voice source features for cognitive load classification	claim
1161	109	Downlink multicell cooperative transmission with imperfect CSI sharing	claim
1162	108	Efficient coherent phase quantization for audio watermarking	claim
1163	108	Target localization with NLOS circularly reflected AOAS	claim
1164	108	Ultra-fast tracking based on zero-shift points	claim
1165	108	Estimating pressure and volume velocity in the ear canal for insert headphones	claim
1166	108	Simulating room impulse responses for spherical microphone arrays	claim
1167	108	A Kalman-like algorithm with no requirements for noise and initial conditions	claim
1168	108	Robust sound field reproduction integrating multi-point sound field control and wave field synthesis	claim
1169	108	Equiangular tight frame fingerprinting codes	claim
1170	108	The least-mean-magnitude-phase algorithm with applications to communications systems	claim
1171	108	A channel-blind system for speaker verification	claim
1172	108	Multiple pitch identification using cochlear-like frequency capture and harmonic grouping	claim
1173	108	Rank-deficient quadratic-form maximization over M-phase alphabet: Polynomial-complexity solvability and algorithmic developments	claim
1174	107	Modeling microstructure noise using Hawkes processes	claim
1175	107	Time-frequency segmentation of bird song in noisy acoustic environments	claim
1176	107	An adaptive approach to abnormal heart sound segmentation	claim
1177	107	Heterogeneous multiprocessor mapping for real-time streaming systems	claim
1178	107	Secure video processing: Problems and challenges	claim
1179	107	Optimum chip pulse shape design for timing synchronization	claim
1180	106	MAP-based estimation of the parameters of non-stationary Gaussian processes from noisy observations	claim
1181	106	Malicious data attack on real-time electricity market	claim
1182	106	Subsequence similarity language models	claim
1183	106	Periodic CRB for non-Bayesian parameter estimation	claim
1184	106	A hierarchical static-dynamic framework for emotion classification	claim
1185	106	Estimation of space-time varying channels using signal subspace projection and soft information	claim
1186	106	Directionality-based speech enhancement for hearing aids	claim
1187	106	Efficient lattice-reduction-aided MMSE decision-feedback equalization	claim
1188	106	An efficient peak frequency estimator for product high-order ambiguity function	claim
1189	106	Exploiting an interplay between norms to analyze scalar quantization schemes	claim
1190	105	Rate-distortion improvement of directional wavelets by megablocking	claim
1191	105	Structured precision modelling with Cholesky Basis Superposition for speech recognition	claim
1192	105	A robust artificial noise aided transmit design for MISO secrecy	claim
1193	105	Third order widely non linear Volterra MVDR beamforming	claim
1194	105	Improving the performance of DSP systems for MIMO processing	claim
1195	105	Nonlinear properties of snoring sounds	claim
1196	105	Function of Phase-Distortion for glottal model estimation	claim
1197	105	Mitigation of reverberation on speaker identification via homomorphic filtering of the linear prediction residual	claim
1198	105	Shout detection in noise	claim
1199	104	The Bayesian inference of phase	claim
1200	104	The grouped two-sided orthogonal Procrustes problem	claim
1201	104	A generalized design method for directivity patterns of spherical microphone arrays	claim
1202	104	Robust speech representation of voiced sounds based on synchrony determination with PLLs	claim
1203	104	Pseudo Maximum Likelihood Estimations of ballistic missile precession frequency	claim
1204	103	A state space model for online polyphonic audio-score alignment	claim
1205	103	Trading off communications bandwidth with accuracy in adaptive diffusion networks	claim
1206	103	Relative proportionate NLMS: Improving convergence for acoustic channel identification	claim
1207	103	Localized filtering for artifact removal in compressed images	claim
1208	103	Spatial sound reproduction systems using higher order loudspeakers	claim
1209	102	Social norm and long-run learning in peer-to-peer networks	claim
1210	102	Blind beamformer for constant modulus signals based on relevance vector machine	claim
1211	102	Neonatal seizure detection using blind multichannel information fusion	claim
1212	102	A single-port non-parametric model of turn-taking in multi-party conversation	claim
1213	102	Probabilistic distance SVM with Hellinger-Exponential Kernel for sound event classification	claim
1214	102	Spatial and temporal power adaptation for space-time coded MIMO systems with imperfect csi	claim
1215	102	Interference self-mitigating beamforming for the K-user MIMO IC	claim
1216	101	Extended array method for detection and direction finding of a weak linear frequency modulated signal	claim
1217	101	Single-symbol decodable distributed STBC for two-path successive relay networks	claim
1218	101	Front-end feature transforms with context filtering for speaker adaptation	claim
1219	100	Multiple-channel detection of a Gaussian time series over frequency-flat channels	claim
1220	100	Sampling and reconstructing diffusion fields with localized sources	claim
1221	100	Image filtering: Potential efficiency and current problems	claim
1222	100	An approach to sequential grouping in cochannel speech	claim
1223	100	A multiwindow time-frequency approach based on the concepts of robust estimate theory	claim
1224	100	A dynamic approach to the selection of high order n-grams in phonotactic language recognition	claim
1225	100	A novel probe processing method for underwater communication by passive-phase conjugation	claim
1226	100	Time-lagged Directed Information	claim
1227	100	A high throughput parallel AVC/H.264 context-based adaptive binary arithmetic decoder	claim
1228	100	Tonal context labeling using quantized F0 symbols for improving tone correctness in average-voice-based speech synthesis	claim
1229	99	Adaptive MIMO detection algorithm by jointly exploiting the properties of signal and channel	claim
1230	99	Bio-inspired swarming models for decentralized radio access incorporating random links and quantized communications	claim
1231	99	Uncovering elements of style	claim
1232	98	Quadrature approximation properties of the spiral-phase quadrature transform	claim
1233	98	Bird species recognition combining acoustic and sequence modeling	claim
1234	98	Joint dictionary learning and topic modeling for image clustering	claim
1235	98	Collaborative system for signal processing education	claim
1236	97	MIMO radar for direction finding with exploitation of time-frequency representations	claim
1237	97	Multi-stage infrared stationary human detection	claim
1238	97	On the use of ideal binary masks for improving phonetic classification	claim
1239	96	Fairness and throughput enhancing user-combining scheme based on Superposition Coding for a wireless relay system	claim
1240	96	Joint analysis of f0 and speech rate with Functional Data Analysis	claim
1241	96	A novel multi-band spectral subtraction method based on phase modification and magnitude compensation	claim
1242	96	Efficient protocols for secure adaptive filtering	claim
1243	96	Discriminative simplification of mixture models	claim
1244	96	Evaluation of objective measures for quality assessment of reverberant speech	claim
1245	96	Reduced-hardware digital filter design via joint quantization and multiple constant multiplication optimization	claim
1246	96	Point process MCMC for sequential music transcription	claim
1247	96	Robust binary least squares: Relaxations and algorithms	claim
1248	95	Decision tree-based context clustering based on cross validation and hierarchical priors	claim
1249	95	Sound morphing by feature interpolation	claim
1250	95	Cooperative prey herding based on diffusion adaptation	claim
1251	95	Sparse spectral factorization: Unicity and reconstruction algorithms	claim
1252	95	Fast physical object identification based on unclonable features and soft fingerprinting	claim
1253	95	PAC-Bayesian approach for minimization of phoneme error rate	claim
1254	95	Emergence of rationality amongst simple nodes performing adaptive filtering	claim
1255	95	Integer fast lapped orthogonal transform based on direct-lifting of dcts for lossless-to-lossy image coding	claim
1256	94	Tone and pitch accent classification using auditory attention cues	claim
1257	94	Characterizing physical-layer secrecy with unknown eavesdropper locations and channels	claim
1258	94	An effective foreground/background segmentation approach for bootstrapping video sequences	claim
1259	94	2D Markovian SS codes flatten time-frequency distribution of signals in asynchronous Gabor division CDMA systems	claim
1260	93	A new trellis representation for source-channel rate allocation	claim
1261	93	Multi-stream spectro-temporal and cepstral features based on data-driven hierarchical phoneme clusters	claim
1262	93	A frame mapping based HMM approach to cross-lingual voice transformation	claim
1263	93	Real-time measurement system for acquiring murine left ventricular pressure-volume relations	claim
1264	93	Estimating note intensities in music recordings	claim
1265	92	Least squares approximation and polyphase decomposition for pipelining recursive filters	claim
1266	92	Clustering and suppression of transient noise in speech signals using diffusion maps	claim
1267	91	Weber's law-based side-informed data hiding	claim
1268	91	Efficiency evaluation and orthogonal basis determination in functional HRTF modeling	claim
1269	91	Difference image extrapolation for spectral completion in inter-sequence error concealment	claim
1270	91	Log spectra enhancement using speaker dependent priors for speaker verification	claim
1271	90	Seismic waves estimation and wave field decomposition with factor graphs	claim
1272	90	Speech synthesis using HMM based diphone inventory encoding for low-resource devices	claim
1273	90	Bound-ratio minimization of filter bank frames by periodic precoding	claim
1274	90	Estimation of the noise correlation matrix	claim
1275	90	Theoretical analysis of musical noise in Wiener filtering family via higher-order statistics	claim
1276	90	Single target tracking using vector magnetometers	claim
1277	89	Detection diversity of multiantenna spectrum sensors	claim
1278	89	Iterative PCA for population structure analysis	claim
1279	89	A machine learning based approach to weather parameter estimation in Doppler weather radar	claim
1280	88	Optimization of the antenna array geometry based on a Bayesian DOA estimation criterion	claim
1281	88	F0 range and peak alignment across speakers and emotions	claim
1282	87	Template-based methods for sentence generation and speech synthesis	claim
1283	87	Extended Viterbi algorithm for optimized word HMMS	claim
1284	87	A multi-dimensional approach to predicting speech quality using a physiologically motivated model of the Cochlea	claim
1285	86	Multi-sensor PHD: Construction and implementation by space partitioning	claim
1286	86	STFT-based denoising of elastograms	claim
1287	86	Forensic voice comparison with secular shibboleths - A hybrid fused gmm-multivariate likelihood ratio-based approach using alveolo-palatal fricative cepstral spectra	claim
1288	86	Iterative FDE for asynchronous single-carrier multiuser systems	claim
1289	86	Broadband direction estimation method utilizing combined pressure and energy gradients from optimized microphone array	claim
1290	86	CLT for eigen-inference methods in cognitive radios	claim
1291	85	Time-evolving modeling of social networks	claim
1292	85	Scale-dependent analysis of Ionosphere fluctuations	claim
1293	85	An analytic approach in joint delay and Doppler estimation using copula	claim
1294	85	Effect of the visual signal structure on Steady-State Visual Evoked Potentials detection	claim
1295	85	Integrating articulatory features using Kullback-Leibler divergence based acoustic model for phoneme recognition	claim
1296	85	Variability regularization in large-margin classification	claim
1297	84	Game-theoretic resource allocation in relay-assisted DS/CDMA systems with successive interference cancellation	claim
1298	84	A passive coupling matrix design for improved resolution small aperture direction finding	claim
1299	84	The role of GTD in optimizing biorthogonal filter banks	claim
1300	83	Zero leakage quantization scheme for biometric verification	claim
1301	83	The meaning of interior tomography	claim
1302	83	Robust direction estimation of UWB sources in Laguerre-Gauss beamspaces	claim
1303	82	Subspace-based DOA estimation using Fractional Lower Order statistics	claim
1304	82	ML decoding in decode-and-forward based cooperative communication system	claim
1305	82	Frequency domain compensation of spurious sidebands in A/D circuits	claim
1306	82	Closed-form expressions vs. BIC: A comparison for speaker clustering	claim
1307	81	A-Functions: A generalization of Extended Baum-Welch transformations to convex optimization	claim
1308	81	When codeword frequency meets geographical location	claim
1309	81	FOR/SOR video coding with super macroblock and inter-frame stripe prediction	claim
1310	81	Timing adjustment techniques to mitigate interference between multiple nodes in OFDMA mesh networks	claim
1311	81	Fast orthogonal decomposition of Volterra cubic kernels using oblique unfolding	claim
1312	81	Augmented complex matrix factorisation	claim
1313	81	Signal-based segmentation of human locomotion using embedded sensor network	claim
1314	80	A data-driven post-filter design based on spatially and temporally smoothed a priori SNR	claim
1315	80	Performance limits of LMS-based adaptive networks	claim
1316	80	Simultaneous SDR optimality via a joint matrix decomposition	claim
1317	80	Improvement to esprit-type frequency estimators via reducing data redundancy	claim
1318	78	Evaluation of two speech and noise estimation methods for the assessment of nonlinear hearing aids	claim
1319	78	Odor stimulus inference based on neural spike signal in rats	claim
1320	78	Open-set speaker identification in broadcast news	claim
1321	77	Angular regularization of vector-valued signals	claim
1322	77	Efficient range extrapolation of head-related impulse responses by wave field synthesis techniques	claim
1323	76	Automatic music tagging via PARAFAC2	claim
1324	76	Is multiparty computation any good in practice?	claim
1325	76	Processing 'yup!' and other short utterances in interactive speech	claim
1326	75	Correlogram template matching for time-delay estimation	claim
1327	75	Industrial application of acousto-ultrasonic signal quality and robust time-of-flight estimation for anisotropic materials	claim
1328	75	A robust estimator and detector of circularity of complex signals	claim
1329	75	Information-theoretic analysis of desynchronization invariant object identification	claim
1330	74	Dissolve detection in abstract video contents	claim
1331	73	Focuss is a convex-concave procedure	claim
1332	73	Video aesthetic quality assessment by combining semantically independent and dependent features	claim
1333	72	G.711.1 Annex D and G.722 Annex B - New ITU-T superwideband codecs	claim
1334	72	Novel beamformers for Multiple Correlated brain source localization and reconstruction	claim
1335	72	Discriminative training for full covariance models	claim
1336	71	A subject-independent acoustic-to-articulatory inversion	claim
1337	71	Non-stationary feature extraction for automatic speech recognition	claim
1338	70	On synthesizing cross ambiguity functions	claim
1339	65	A collaborative filtering approach for quasi-brain-death EEG analysis	claim
1340	65	Enhanced long-term predictor for Unified Speech and Audio Coding	claim
1341	63	Pinna sensitivity patterns reveal reflecting and diffracting surfaces that generate the first spectral notch in the front median plane	claim
1342	63	Simulation of hearing loss using compressive gammachirp auditory filters	claim
1343	63	Direct processing of mpeg audio using companding and BFP techniques	claim
1344	62	Steady-state analysis of the NLMS algorithm with reusing coefficient vector and a method for improving its performance	claim
1345	62	Deconvolution of neuronal signal from hemodynamic response	claim
1346	62	The cost of JPEG compression anti-forensics	claim
1347	61	Overlapped speech detection using long-term spectro-temporal similarity in stereo recording	claim
1348	60	Speech recognition modeling advances for mobile voice search	claim
1349	60	On-line Memory-Based Parametric Equalization to multimodal training conditions	claim
1350	60	Orthogonal wavelet division multiplexing for wideband time-varying channels	claim
1351	60	Empirical evidence against CAPM: Relating alphas and returns to betas	claim
1352	58	Automatic estimation of the second subglottal resonance from natural speech	claim
1353	57	Detection of sinusoidal signals in noise by probabilistic modelling of the spectral magnitude shape and phase continuity	claim
1354	57	Detection and segmentation of fmcw radar signals based on the chirplet transform	claim
1355	50	Towards reduced false-alarms using cohorts	claim
1356	49	Speech-based identification of social groups in a single accent of British English by humans and computers	claim
1357	45	A unified framework for GLRT-based spectrum sensing of signals with covariance matrices with known eigenvalue multiplicities	claim

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