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ICASSP 2011 Speech Acoustics

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ICASSP
2011
IEEE

438views Signal Processing» more ICASSP 2011»

Non-flat clustering whith alpha-divergences

1 years 2 months ago

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

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