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ICASSP 2010 Speech Acoustics

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ICASSP
2010
IEEE

390views Signal Processing» more ICASSP 2010»

Human detection in images via L1-norm Minimization Learning

13 years 10 months ago

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In recent years, sparse representation originating from signal compressed sensing theory has attracted increasing interest in computer vision research community. However, to our b...

Ran Xu, Baochang Zhang, Qixiang Ye, Jianbin Jiao

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Id	Views	Title	Status

1	390	Human detection in images via L1-norm Minimization Learning	claim
2	356	A new sparsity-enabled signal separation method based on signal resonance	claim
3	330	Evaluation of Distance Based Amplitude panning for spatial audio	claim
4	297	Modulated wideband converter with non-ideal lowpass filters	claim
5	281	Fast structural similarity index algorithm	claim
6	278	Hierarchical dictionary learning for invariant classification	claim
7	274	Geometric tampering estimation by means of a SIFT-based forensic analysis
8	273	Hierarchical Gaussian Mixture Model	claim
9	267	A compressive sensing image compression algorithm using quantized DCT and noiselet information	claim
10	265	Clustering disjoint subspaces via sparse representation	claim
11	259	An adaptive level of detail approach to nonlinear estimation	claim
12	254	Multipass strategies for improving accuracy in a voice search application	claim
13	251	Adaptive beam tracking for interference alignment for multiuser time-varying MIMO interference channels	claim
14	241	No-reference image sharpness assessment based on local phase coherence measurement	claim
15	239	Towards multi-speaker unsupervised speech pattern discovery	claim
16	239	Bayesian compressive sensing for phonetic classification	claim
17	238	An estimation method for the relative phase parameters of complex wavelet coefficients in noise	claim
18	238	Singing information processing based on singing voice modeling	claim
19	238	Speech enhancement with sparse coding in learned dictionaries	claim
20	237	Unsupervised knowledge acquisition for Extracting Named Entities from speech	claim
21	237	Background music identification through content filtering and min-hash matching	claim
22	233	Vocabulary and language model adaptation using just one speech file	claim
23	231	Automated ECG profiling and beat classification	claim
24	228	Anti-forensics of JPEG compression	claim
25	227	Designing the Wiener post-filter for diffuse noise suppression using imaginary parts of inter-channel cross-spectra	claim
26	225	Toward signal processing theory for graphs and non-Euclidean data	claim
27	224	Distributed learning in cognitive radio networks: Multi-armed bandit with distributed multiple players	claim
28	224	Action change detection in video by covariance matching of silhouette tunnels	claim
29	223	Investigations into prosodic syllable contour features for speaker recognition	claim
30	221	Searching with expectations	claim
31	220	A distributed psycho-visually motivated Canny edge detector	claim
32	216	Quaternionic wavelets for texture classification	claim
33	216	Flexible complex ICA of fMRI data	claim
34	215	Context-adaptive pixel based prediction for intra frame encoding	claim
35	214	Implementation of cooperative communications using software defined radios	claim
36	214	Algorithms for robust linear regression by exploiting the connection to sparse signal recovery	claim
37	213	Buffer management for multi-application image processing on multi-core platforms: Analysis and case study	claim
38	213	A nonnegative sparsity induced similarity measure with application to cluster analysis of spam images	claim
39	213	Fast total variation image restoration with parameter estimation using bayesian inference	claim
40	210	Gradient descent approach for secure localization in resource constrained wireless sensor networks	claim
41	209	Optimal phase-shifter design to cancel RF interference in multi-antenna systems	claim
42	206	Camera-based clear path detection	claim
43	206	Visual saliency for automatic target detection, boundary detection, and image quality assessment	claim
44	205	Content-aware H.264 encoding for traffic video tracking applications	claim
45	204	MMSE based noise PSD tracking with low complexity	claim
46	204	On expectation maximization based channel and noise estimation beyond the vector Taylor series expansion	claim
47	204	Semi-Supervised Hyperspectral Unmixing via the Weighted Lasso	claim
48	203	Tracking encrypted VoIP calls via robust hashing of network flows	claim
49	203	Bandwidth-intensive FPGA architecture for multi-dimensional DFT	claim
50	202	Acoustic model adaptation via Linear Spline Interpolation for robust speech recognition	claim
51	200	Kronecker product matrices for compressive sensing	claim
52	200	Shape matching based on graph alignment using hidden Markov models	claim
53	200	A sparse component model of source signals and its application to blind source separation	claim
54	199	A weighted discriminative approach for image denoising with overcomplete representations	claim
55	198	Near-field adaptive beamforming and source localization in the spacetime frequency domain	claim
56	198	Recognition of phonemes and words in singing	claim
57	197	Subpixel motion estimation without interpolation	claim
58	197	Bridge detection and robust geodesics estimation via random walks	claim
59	196	Consensus in correlated random topologies: Weights for finite time horizon	claim
60	195	Adaptive compressed sensing - A new class of self-organizing coding models for neuroscience	claim
61	195	Real-time content adaptive contrast enhancement for see-through fog and rain	claim
62	194	Fundamental limits of image denoising: Are we there yet?	claim
63	194	Compressive detection for wide-band spectrum sensing	claim
64	192	A convex relaxation for approximate maximum-likelihood 2D source localization from range measurements	claim
65	191	Supervised topic model for automatic image annotation	claim
66	191	Sound source separation in monaural music signals using excitation-filter model and em algorithm	claim
67	191	Audio forensics from acoustic reverberation	claim
68	190	Acoustic front-end optimization for bird species recognition	claim
69	190	The effect of lattice pruning on MMIE training	claim
70	190	Disparity search range estimation: Enforcing temporal consistency	claim
71	190	Approximate nearest neighbors using sparse representations	claim
72	189	Graph based event detection from realistic videos using weak feature correspondence	claim
73	189	A new method for kurtosis maximization and source separation	claim
74	189	Graph-spectrum-based neural spike features for stereotrodes and tetrodes	claim
75	189	Energy-based multi-speaker voice activity detection with an ad hoc microphone array	claim
76	188	Feedback reduction in MIMO broadcast channels with LMMSE receivers	claim
77	188	The IBM 2008 GALE Arabic speech transcription system	claim
78	188	Learning with synthesized speech for automatic emotion recognition	claim
79	187	Multi-View Object Detection by Classifier Interpolation	claim
80	187	Joint bandwidth and power allocation in wireless multi-user decode-and-forward relay networks	claim
81	187	Blind rhythmic source separation: Nonnegativity and repeatability	claim
82	186	An adaptive real-time descreening method based on SVM and improved SUSAN filter	claim
83	184	An alternate approach to adaptive beamforming using SRP-PHAT	claim
84	183	Temporally constrained SCA with applications to EEG data	claim
85	182	A statistical model-based double-talk detection incorporating soft decision	claim
86	182	A practical forwarding scheme for wireless relay channels based on the quantization of log-likelihood ratios	claim
87	182	Search-Based Automatic Image Annotation via Flickr Photos Using Tag Expansion	claim
88	182	Identification of linear systems in canonical form through an EM framework	claim
89	182	Psychoacoustically motivated, frequency dependent Tikhonov regularization for soundfield parametrization	claim
90	181	Distributed reconstruction of time-varying spatial fields based on consensus propagation	claim
91	181	Two novel compressed-sensing algorithms for NBI detection in OFDM systems	claim
92	181	A new penalty term for the BIC with respect to speaker diarization	claim
93	181	Modified hierarchical clustering for sparse component analysis	claim
94	180	On-the-fly video genre classification by combination of audio features	claim
95	179	Multi-modal analysis of dance performances for music-driven choreography synthesis	claim
96	179	Automatic state discovery for unstructured audio scene classification	claim
97	179	Variational nonparametric Bayesian Hidden Markov Model	claim
98	179	Analysis of phone posterior feature space exploiting class-specific sparsity and MLP-based similarity measure	claim
99	179	A comparison of approximate Viterbi techniques and particle filtering for data estimation in digital communications	claim
100	178	Large margin estimation of n-gram language models for speech recognition via linear programming	claim
101	178	Limited feedback with joint CSI quantization for multicell cooperative generalized eigenvector beamforming	claim
102	178	A novel, fast, and complete 3D segmentation of vertebral bones	claim
103	177	NC node selection game in collaborative streaming systems	claim
104	177	Characterization of movie genre based on music score	claim
105	177	Optimization of a maximum entropy criterion for 2D Nuclear Magnetic Resonance reconstruction	claim
106	176	Speech/Non-Speech Detection in Meetings from Automatically Extracted low Resolution Visual Features	claim
107	176	Towards objectifying information hiding	claim
108	175	Cyclic tempogram - A mid-level tempo representation for musicsignals	claim
109	175	Alpha-integration of multiple evidence	claim
110	175	Perceptual audio features for unsupervised key-phrase detection	claim
111	175	An efficient particle filtering technique on the Grassmann manifold	claim
112	175	Using reed-muller sequences as deterministic compressed sensing matrices for image reconstruction	claim
113	174	Direct importance estimation with probabilistic principal component analyzers	claim
114	174	Collaborative spectrum sensing from sparse observations using matrix completion for cognitive radio networks	claim
115	173	Concentration of measure for block diagonal measurement matrices	claim
116	173	Partial sequence matching using an Unbounded Dynamic Time Warping algorithm	claim
117	173	Music dereverberation using harmonic structure source model and Wiener filter	claim
118	173	Wideband spectral estimation from compressed measurements exploiting spectral a priori information in Cognitive Radio systems	claim
119	173	VTLN adaptation for statistical speech synthesis	claim
120	173	Learning in Gaussian Markov random fields	claim
121	173	Loquendo-Politecnico di Torino system for the 2009 NIST Language Recognition Evaluation	claim
122	172	An improved consensus-like method for Minimum Bayes Risk decoding and lattice combination	claim
123	172	Design of sparse filters for channel shortening	claim
124	172	Framework for cross-language automatic phonetic segmentation	claim
125	172	Detection-based speech recognition with sparse point process models	claim
126	171	Particle filter adaptation for distributed sensors via set membership	claim
127	171	Learning deep rhetorical structure for extractive speech summarization	claim
128	171	On the use of machine translation for spoken language understanding portability	claim
129	170	Gradient Polytope Faces Pursuit for large scale sparse recovery problems	claim
130	170	Convergence analysis of consensus-based distributed clustering	claim
131	170	Robust critical data recovery for MPEG-4 AAC encoded bitstreams	claim
132	170	Ergodic stochastic optimization algorithms for wireless communication and networking	claim
133	170	Quality assessment for listening-room compensation algorithms	claim
134	169	Morphology-based and sub-word language modeling for Turkish speech recognition	claim
135	169	Evaluation of semantic role labeling and dependency parsing of automatic speech recognition output	claim
136	169	3D shape estimation from silhouettes using mean-shift	claim
137	169	Beyond Doddington menagerie, a first step towards	claim
138	169	2-D two-fold symmetric circular shaped filter design with homomorphic processing application	claim
139	169	A comparison of approaches for modeling prosodic features in speaker recognition	claim
140	168	Sound enhancement using sparse approximation with speclets	claim
141	168	Discriminative template extraction for direct modeling	claim
142	168	Fishervioce: A discriminant subspace framework for speaker recognition	claim
143	167	A novel watermarking scheme with compensation in bit-stream domain for H.264/AVC	claim
144	167	A H.264/AVC video database for the evaluation of quality metrics	claim
145	167	A HMM-based speech synthesis system using a new glottal source and vocal-tract separation method	claim
146	166	A kernel-approach for estimating the position of moving objects	claim
147	166	Wideband noise suppression supported by artificial bandwidth extension techniques	claim
148	166	Visual localization and segmentation based on foreground/background modeling	claim
149	166	Adaptive search for sparse targets with informative priors	claim
150	166	Independent subspace analysis with prior information for fMRI data	claim
151	165	Graph-based regularization for spherical signal interpolation	claim
152	165	Boosted binary features for noise-robust speaker verification	claim
153	165	Distributed bearing estimation via matrix completion	claim
154	165	An online quasi-Newton algorithm for blind SIMO identification	claim
155	165	A low complexity speck-based codec for multispectral fluorescence microscopic images	claim
156	164	Under-determined convolutive blind source separation using spatial covariance models	claim
157	164	Partial clustering using a time-varying frequency model for singing voice detection	claim
158	164	Improved Internet traffic analysis via optimized sampling	claim
159	163	Discriminative training methods for language models using conditional entropy criteria	claim
160	163	Iterated smoothing for accelerated gradient convex minimization in signal processing	claim
161	163	Semantic confidence calibration for spoken dialog applications	claim
162	163	Swift: Scalable weighted iterative sampling for flow cytometry clustering	claim
163	163	Estimation of a white Gaussian noise in the Short Time Fourier Transform based on the spectral kurtosis of the minimal statistics: Application to underwater noise	claim
164	163	Enhanced trellis based vector quantization for coordinated beamforming	claim
165	163	Search error risk minimization in Viterbi beam search for speech recognition	claim
166	162	Distributed Lasso for in-network linear regression	claim
167	162	Towards jointly optimal spatial prediction and adaptive transform in video/image coding	claim
168	162	Model-based dereverberation in the logmelspec domain for robust distant-talking speech recognition	claim
169	161	Statistical modeling of co-channel interference in a field of Poisson distributed interferers	claim
170	161	Best-effort cooperative communication without dedicated relays	claim
171	161	Statistical approach to enhancing esophageal speech based on Gaussian mixture models	claim
172	161	Phase correction and denoising for ICA of complex FMRI data	claim
173	160	Evolutionary spectral clustering with adaptive forgetting factor	claim
174	160	Robust regression using sparse learning for high dimensional parameter estimation problems	claim
175	160	Temporal motion smoothness measurement for reduced-reference video quality assessment	claim
176	160	Rapid integration of Parts of Speech information to improve reordering model for English-Farsi Speech to Speech Translation	claim
177	160	Performance analysis of IPNLMS for identification of time-varying systems	claim
178	159	Approaches to automatic lexicon learning with limited training examples	claim
179	159	Multiple sequence alignment based bootstrapping for improved incremental word learning	claim
180	158	Multiple Selection Approximation for improved spatio-temporal prediction in video coding	claim
181	158	Evaluation of random-projection-based feature combination on speech recognition	claim
182	158	On the tracking performance of combinations of least mean squares and recursive least squares adaptive filters	claim
183	158	Comparison of modulation features for phoneme recognition	claim
184	158	Effect of speaking style and speaking rate on formant contours	claim
185	158	Symmetrical EEG/FMRI fusion with spatially adaptive priors using variational distribution approximation	claim
186	157	Fixed-budget kernel recursive least-squares	claim
187	157	Adaptive spectrum sensing for agile cognitive radios	claim
188	157	Speaker identification by combining MFCC and phase information in noisy environments	claim
189	157	Synthesis of filled pauses based on a disfluent speech model
190	157	Fast signal analysis and decomposition on graphs using the Sparse Matrix Transform	claim
191	157	Fast likelihood computation using hierarchical Gaussian shortlists	claim
192	157	Towards effective singing voice extraction from stereophonic recordings	claim
193	157	Building pair-wise visual word tree for efficent image re-ranking	claim
194	156	Image-quality prediction of synthetic aperture sonar imagery	claim
195	156	An auction-based framework for multimedia streaming over cognitive radio networks	claim
196	156	An L1 criterion for dictionary learning by subspace identification	claim
197	156	Noise robust exemplar-based connected digit recognition	claim
198	156	CBCD based on color features and landmark MDS-assisted distance estimation	claim
199	156	Disambiguation in multidimensional tracking of multiple acoustic sources using a Gaussian likelihood criterion	claim
200	155	Maximizing the lifetime of clusters with Slepian-Wolf coding	claim
201	155	Performance analysis for blind identification of acoustic channels	claim
202	155	Parametric dictionary learning using steepest descent	claim
203	155	GHT based implementation of the expectation maximization for mixtures of multi-Gaussians and its applications to video tracking	claim
204	154	Aspect-model-based reference speaker weighting	claim
205	154	Histogram equalization and noise masking for robust speech recognition	claim
206	154	Summarization- and learning-based approaches to information distillation	claim
207	154	Detecting local semantic concepts in environmental sounds using Markov model based clustering	claim
208	154	Noise-to-mask ratio minimization by weighted non-negative matrix factorization	claim
209	154	Asymptotic analysis of the Huberized LASSO estimator	claim
210	154	Exploring statistical properties for semantic annotation: sparse distributed and convergent assumptions for keywords	claim
211	154	Physical layer algorithm and hardware verification of MIMO relays using cooperative partial detection	claim
212	154	Feature-based extraction of plucking and expression styles of the electric bass guitar	claim
213	153	Texas Hold 'Em algorithms for distributed compressive sensing	claim
214	153	Fast semi-supervised image segmentation by novelty selection	claim
215	152	Generalized Discrete Fourier Transform with optimum correlations	claim
216	152	HMM-based sequence-to-frame mapping for voice conversion	claim
217	152	Power law discounting for n-gram language models	claim
218	152	Acceleration of sequence kernel computation for real-time speaker identification	claim
219	152	Multilingual acoustic modeling for speech recognition based on subspace Gaussian Mixture Models	claim
220	152	Transcription-based video genre classification	claim
221	152	A robust minimum volume enclosing simplex algorithm for hyperspectral unmixing	claim
222	151	An analytical approach to 2.5D sound field reproduction employing linear distributions of non-omnidirectional loudspeakers	claim
223	151	Guaranteed robust distributed estimation in a network of sensors	claim
224	151	Efficient online learning with individual learning-rates for phoneme sequence recognition	claim
225	151	Direction-of-arrival and spatial signature estimation in antenna arrays with pairwise sensor calibration	claim
226	151	Multichannel-compressive estimation of doubly selective channels in MIMO-OFDM systems: Exploiting and enhancing joint sparsity	claim
227	150	Optimizing spectral subtraction and wiener filtering for robust speech recognition in reverberant and noisy conditions	claim
228	150	Automatic matched filter recovery via the audio camera	claim
229	150	Automatic detection of well sampled images via a new ringing measure	claim
230	150	Green modulation in dense Wireless Sensor Networks	claim
231	150	Error resilient video multicast using Randomized Distributed Space Time Codes	claim
232	150	HMM-based separation of acoustic transfer function for single-channel sound source localization	claim
233	150	Feature extraction for robust speech recognition based on maximizing the sharpness of the power distribution and on power flooring	claim
234	149	Baffled circular loudspeaker array with broadband high directivity	claim
235	149	High frame rate Motion Compensated Frame Interpolation in High-Definition video processing	claim
236	149	Model-level data-driven sub-units for signs in videos of continuous Sign Language	claim
237	149	Energy efficient lossy transmission over sensor networks with feedback	claim
238	148	On unbiased estimation of sparse vectors corrupted by Gaussian noise	claim
239	148	Empirical quantization for sparse sampling systems	claim
240	148	Multiantenna spectrum sensing: Detection of spatial correlation among time-series with unknown spectra	claim
241	148	Tuning phone decoders for language identification	claim
242	148	Compressive sensing and differential image-motion estimation	claim
243	147	Matching canvas weave patterns from processing x-ray images of master paintings	claim
244	147	Audio fingerprinting to identify multiple videos of an event	claim
245	147	Training a support vector machine to classify signals in a real environment given clean training data	claim
246	147	A survey of implementation efforts and experimental design for cooperative communications	claim
247	147	Multi-class SVM optimization using MCE training with application to topic identification	claim
248	147	Seamlets: Content-aware nonlinear wavelet transform	claim
249	146	Song-level multi-pitch tracking by heavily constrained clustering	claim
250	146	Linear transceiver design in a multiuser MIMO system with quantized channel state information	claim
251	146	Privacy and security of features extracted from minutiae aggregates	claim
252	146	Approximate eigenvalue decomposition of para-Hermitian systems through successive FIR paraunitary transformations	claim
253	146	Random attributed graphs for statistical inference from content and context	claim
254	145	A minimax approach to Bayesian estimation with partial knowledge of the observation model	claim
255	145	Multiple frequency-hopping signal estimation via sparse regression	claim
256	145	Removing illumination artifacts from face images using the nuisance attribute projection	claim
257	145	Flexcode - flexible audio coding	claim
258	145	Language model adaptation using Random Forests	claim
259	145	Language recognition using deep-structured conditional random fields	claim
260	145	Noisy speech enhancement based on prior knowledge about spectral envelope and harmonic structure	claim
261	145	Using online model comparison in the Variational Bayes framework for online unsupervised Voice Activity Detection	claim
262	145	High order motion interpolation for side information improvement in DVC	claim
263	145	Investigations on ensemble based unsupervised adaptation methods	claim
264	145	Forensic estimation and reconstruction of a contrast enhancement mapping	claim
265	145	A supervisory approach to semi-supervised clustering	claim
266	144	Syntactic and sub-lexical features for Turkish discriminative language models	claim
267	144	Source-domain adaptive filtering for MIMO systems with application to acoustic echo cancellation	claim
268	144	Statistics of natural image distortions	claim
269	144	Simultaneous search for all modes in multilinear models	claim
270	144	A digital microphone array for distant speech recognition	claim
271	144	Discriminative HMMS, log-linear models, and CRFS: What is the difference?	claim
272	144	Voice source estimation for artificial bandwidth extension of telephone speech	claim
273	144	Sub-Nyquist processing with the modulated wideband converter	claim
274	143	Robust spectro-temporal features based on autoregressive models of Hilbert envelopes	claim
275	143	Paraphrase detection on SMS messages in automobiles	claim
276	143	Multi-object filtering from image sequence without detection	claim
277	143	Source-relay optimization for a two-way MIMO relay system	claim
278	143	Automatic disfluency removal for improving spoken language translation	claim
279	143	Audio-based nonlinear video diffusion	claim
280	142	Using n-best recognition output for extractive summarization and keyword extraction in meeting speech	claim
281	142	An exploratory study of manifolds of emotional speech	claim
282	142	Using cross-decoder phone coocurrences in phonotactic language recognition	claim
283	142	Improved statistical models for SMT-based speaking style transformation	claim
284	142	Stability analysis of an adaptive Wiener structure	claim
285	142	Unsupervised cross-lingual speaker adaptation for HMM-based speech synthesis	claim
286	141	Reconstructing the full tongue contour from EMA/X-ray microbeam	claim
287	141	Word confidence calibration using a maximum entropy model with constraints on confidence and word distributions	claim
288	141	Simulating dynamic communication systems using the core functional dataflow model	claim
289	141	Multiplicative update rules for nonnegative matrix factorization with co-occurrence constraints	claim
290	141	Maximum-likelihood-based cepstral inverse filtering for blind speech dereverberation	claim
291	141	Hiding information inside structured shapes	claim
292	141	Ranging energy optimization for robust sensor positioning with collaborative anchors	claim
293	141	A new model for Gabor coefficients' magnitude in face recognition	claim
294	141	Convergence behavior of the Active Mask segmentation algorithm	claim
295	140	Time-varying spectrum estimation of uniformly modulated processes by means of surrogate data and empirical mode decomposition	claim
296	140	Comparing the contributions of context and prosody in text-independent dialog act recognition	claim
297	140	High precision frequency estimation for harpsichord tuning classification	claim
298	140	Improving speech recognition by explicit modeling of phone deletions	claim
299	140	Clap your hands! Calibrating spectral subtraction for dereverberation	claim
300	140	Learning from other subjects helps reducing Brain-Computer Interface calibration time	claim
301	140	Robust frequency-based Audio Fingerprinting	claim
302	139	Building a web platform for learning advanced digital communications using a MIMO testbed	claim
303	139	Quantifying EEG synchrony using copulas	claim
304	139	Spatio-angular sharpening for multiview 3D displays	claim
305	139	Speech presence probability estimation based on temporal cepstrum smoothing	claim
306	139	Synthesizing speech from Doppler signals	claim
307	139	Robust similarity metrics between audio signals based on asymmetrical spectral envelope matching	claim
308	139	Structured and incoherent parametric dictionary design	claim
309	139	A partially collapsed Gibbs sampler for parameters with local constraints	claim
310	138	Stochastic pronunciation modelling and soft match for out-of-vocabulary spoken term detection	claim
311	138	Latent-variable decomposition based dereverberation of monaural and multi-channel signals	claim
312	138	Robust speaker identification using an auditory-based feature	claim
313	138	Two-stage spectrum sensing for cognitive radios	claim
314	138	High-resolution glyph-inspection based security system	claim
315	138	Subspace Gaussian Mixture Models for speech recognition	claim
316	138	Classifying laughter and speech using audio-visual feature prediction	claim
317	138	Across-phone variability and diagonal term in joint factor analysis for speaker recognition	claim
318	138	On the use of speaker superfactors for speaker recognition	claim
319	138	Balancing false alarms and hits in Spoken Term Detection	claim
320	138	Video-based fingerprint verification	claim
321	137	Jointly recognizing multi-speaker conversations	claim
322	137	Ultrasound tomography with learned dictionaries	claim
323	137	Latent topic modeling of word vicinity information for speech recognition	claim
324	137	An adaptive initialization method for speaker Diarization based on prosodic features	claim
325	137	Rate distortion optimization for bidirectional scalable motion model	claim
326	137	Towards large scale distributed coding	claim
327	137	Sparse signal recovery in the presence of correlated multiple measurement vectors	claim
328	137	A signal-specific bound for joint tdoa and FDOA estimation and its Use in combining multiple segments	claim
329	136	Autoregressive modeling of voiced speech	claim
330	136	A kernel mean matching approach for environment mismatch compensation in speech recognition	claim
331	136	Informed source separation of underdetermined instantaneous stereo mixtures using source index embedding	claim
332	136	Learning from high-dimensional noisy data via projections onto multi-dimensional ellipsoids	claim
333	136	Hierarchical language modeling for audio events detection in a sports game	claim
334	136	Circulant space-time codes for integration with beamforming	claim
335	136	Mining actor correlations with hierarchical concurrence parsing	claim
336	135	A parallel point-process filter for estimation of goal-directed movements from neural signals	claim
337	135	Revisiting the security of speaker verification systems against imposture using synthetic speech	claim
338	135	Automatic audio tagging using covariate shift adaptation	claim
339	135	Visual emotion recognition using compact facial representations and viseme information	claim
340	135	Structuring a gene network using a multiresolution independence test	claim
341	135	Adaptive anisotropic regularization of deformation fields for non-rigid registration using the morphon framework	claim
342	135	Robust MIMO radar detection for correlated subarrays	claim
343	135	HMM-based pseudo-clean speech synthesis for splice algorithm	claim
344	135	Integrating monaural and binaural analysis for localizing multiple reverberant sound sources	claim
345	135	Reconstruction of sparse signals from distorted randomized measurements	claim
346	135	From flat direct models to segmental CRF models	claim
347	135	Total-variation regularization with bound constraints	claim
348	135	Unidirectional graph-based wavelet transforms for efficient data gathering in sensor networks	claim
349	135	Sectored Random Projections for Cancelable Iris Biometrics	claim
350	134	Low-latency online speaker tracking on the AMI Corpus of meeting conversations	claim
351	134	On the outage and diversity-multiplexing tradeoff of broadcast channels with 1-bit feedback	claim
352	134	Finding emotionally involved speech using implicitly proximity-annotated laughter	claim
353	134	Semi-Supervised Fisher Linear Discriminant (SFLD)	claim
354	134	Leveraging evaluation metric-related training criteria for speech summarization	claim
355	134	Information retrieval methods for automatic speech recognition	claim
356	134	An alternative scanning strategy to detect faces	claim
357	134	Performance of shockwave-based shooter localization under model misspecification	claim
358	133	Coherence-based near-oracle performance guarantees for sparse estimation under Gaussian noise	claim
359	133	Phone recognition using Restricted Boltzmann Machines	claim
360	133	Hypothesis ranking and two-pass approaches for machine translation system combination	claim
361	133	Quantization and compensation in sampled interleaved multi-channel systems	claim
362	132	Cover song detection: From high scores to general classification	claim
363	132	Speaker independent visual-only language identification	claim
364	132	A novel estimation of feature-space MLLR for full-covariance models	claim
365	132	Distance-based discretization of parametric signal manifolds	claim
366	132	On distance reconstruction for sensor network localization	claim
367	132	Speech modeling based on committee-based active learning	claim
368	132	A general formalism for the analysis of distributed algorithms	claim
369	131	Multimodal similarity between musical streams for cover version detection	claim
370	131	Non-parallel training for many-to-many eigenvoice conversion	claim
371	131	Reverberated speech signal separation based on regularized subband feedforward ICA and instantaneous direction of arrival	claim
372	131	A nullspace analysis of the nuclear norm heuristic for rank minimization	claim
373	131	A hierarchical Bayesian model for frame representation	claim
374	131	Simple methods for improving speaker-similarity of HMM-based speech synthesis	claim
375	130	Performance following: Tracking a performance without a score	claim
376	130	A study of several model selection criteria for determining the number of signals	claim
377	130	A short-time objective intelligibility measure for time-frequency weighted noisy speech	claim
378	129	Sparsity-cognizant overlapping co-clustering for behavior inference in social networks	claim
379	129	Evaluation of a stereo audio data hiding method using inter-channel decorrelator polarity	claim
380	129	Hyperspectral target detection from incoherent projections	claim
381	128	Weakly supervised learning with decision trees applied to fisheries acoustics	claim
382	128	Motion estimation from compressed linear measurements	claim
383	128	NMF with time-frequency activations to model non stationary audio events	claim
384	128	Widely and semi-widely linear processing of quaternion vectors	claim
385	128	Learning-based auditory encoding for robust speech recognition	claim
386	128	Optimize the obvious: Automatic call flow generation	claim
387	128	Improved single-channel speech separation using sinusoidal modeling	claim
388	127	Utilizing principal singular vectors for two-dimensional single frequency estimation	claim
389	127	Estimation of frame independent and enhancement components for speech communication over packet networks	claim
390	127	Robust background modeling via standard variance feature	claim
391	127	Efficient weighted-sum-rate maximization for a class of half-duplex cooperative systems	claim
392	127	Statistical hypothesis testing with time-frequency surrogates to check signal stationarity	claim
393	126	Morphological wavelets and the complexity of dyadic trees	claim
394	126	Optimized intrinsic dimension estimator using nearest neighbor graphs	claim
395	126	Sparse Bayesian step-filtering for high-throughput analysis of molecular machine dynamics	claim
396	126	Learning sparse systems at sub-Nyquist rates: A frequency-domain approach	claim
397	126	Enhancing sparsity in linear prediction of speech by iteratively reweighted 1-norm minimization	claim
398	126	L1 regularized room modeling with compact microphone arrays	claim
399	126	A transient analysis for the convex combination of two adaptive filters with transfer of coefficients	claim
400	125	Prior-based vanishing point estimation through global perspective structure matching	claim
401	125	A comparison of nonstaggered compact FDTD schemes for the 3D wave equation	claim
402	125	A comparative study of tonal acoustic features for a symbolic level music-to-score alignment	claim
403	125	A symmetric adaptive algorithm for speeding-up consensus	claim
404	124	Implementing LNS using filtering units of GPUs	claim
405	124	Predicting interruptions in dyadic spoken interactions	claim
406	124	Spoken language translation from parallel speech audio: Simultaneous interpretation as SLT training data	claim
407	123	Statistical Resolution Limit for multiple parameters of interest and for multiple signals	claim
408	123	Compressive sensing of a superposition of pulses	claim
409	123	Speech enhancement in transient noise environment using diffusion filtering	claim
410	123	Exponentially embedded families for multimodal sensor processing	claim
411	122	Nonlinear kernel backprojection for computed tomography	claim
412	122	Sparsity maximization under a quadratic constraint with applications in filter design	claim
413	121	A shape-based framework to segmentation of tongue contours from MRI data	claim
414	121	Voice activity detection using harmonic frequency components in likelihood ratio test	claim
415	121	A System-Identification-Error-Robust Method for equalization of multichannel acoustic systems	claim
416	121	Particle filtering based recovery of noisy GARCH processes	claim
417	121	On linear versus non-linear magnitude-DFT estimators and the influence of super-Gaussian speech priors	claim
418	120	Empirical Type-i filter design for image interpolation	claim
419	120	Optimal gain control for single-carrier communications with uniform quantization at the receiver	claim
420	120	Joint estimation of DOA and speech based on EM beamforming	claim
421	119	Subword-based spoken term detection in audio course lectures	claim
422	119	Outage-optimal transmission in multiuser-MIMO Kronecker channels	claim
423	117	Frame domain signal processing: Framework and applications	claim
424	117	Concave resource allocation problems for interference coupled wireless systems	claim
425	117	Field trial results for LTE-advanced concepts	claim
426	117	Note onset detection using rhythmic structure	claim
427	117	Subjective ratings of instantaneous and gradual transitions from narrowband to wideband active speech	claim
428	117	An improved deviation measure for two-path echo cancellation	claim
429	116	Magnitude spectrum enhancement for robust speech recognition	claim
430	116	Large margin filtering for Signal Sequence Labeling	claim
431	116	Oversampling transmit and receive antenna arrays	claim
432	115	Speaker identification with distant microphone speech	claim
433	115	Energy-aware adaptive OFDM systems	claim
434	114	Voice quality evaluation of various codecs	claim
435	114	Ultrasonic sensing for robust speech recognition	claim
436	114	Stochastic cross-layer resource allocation for wireless networks using orthogonal access: Optimality and delay analysis	claim
437	113	Sparse representation algorithms based on mean-field approximations	claim
438	113	On the Shannon capacity of DNA data embedding	claim
439	113	Multi-style MLP features for BN transcription	claim
440	113	Use of imperfectly segmented nuclei in the classification of histopathology images of breast cancer	claim
441	112	A non-iterative estimator for the concentration parameter of the relative phase distribution	claim
442	112	Optimal delayed decoding of predictively encoded sources	claim
443	112	Robust rate-maximization game under bounded channel uncertainty	claim
444	111	Statistical parametric speech synthesis based on product of experts	claim
445	111	Restoration-based iron oxide particles quantification in MR images	claim
446	111	Incremental partition recombination for efficient tracking of multiple dialog states	claim
447	110	Joint estimate of shape and time-synchronization of a glottal source model by phase flatness	claim
448	110	Decomposing tensors with structured matrix factors reduces to rank-1 approximations	claim
449	108	A union of incoherent spaces model for classification	claim
450	107	Using the Amazon Mechanical Turk for transcription of spoken language	claim
451	106	On the robustness of the multidimensional state coherence transform for solving the permutation problem of frequency-domain ICA	claim
452	105	Rate-distortion optimized transform for intra-frame coding	claim
453	102	A new voice source model based on high-speed imaging and its application to voice source estimation	claim
454	101	Robust IP and UDP-lite header recovery for packetized multimedia transmission	claim
455	101	Spread spectrum for interferometric and magnetic resonance imaging	claim
456	101	Multipoint-to-point and point-to-multipoint space-time network coding	claim
457	100	Statistical inference for single- and multi-band Probabilistic Amplitude Demodulation	claim
458	99	Harmonic variable-size dictionary learning for music source separation	claim
459	92	Preserving the character of perturbations in scaled pitch contours	claim
460	81	Redundant representation for network video streaming using reconstructed P-frames and SP-frames	claim

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