ICASSP 2009 - PDF Preprint Papers

ICASSP 2009 Speech Acoustics

Teaser Browsing | Click Here

Title Browsing | Click Here

Wall of Fame | Most Viewed ICASSP-2009 Paper

click to vote

ICASSP
2009
IEEE

979views Signal Processing» more ICASSP 2009»

Image Recolorization for the Colorblind

2 years 7 months ago

Download sites.google.com

In this paper, we propose a new re-coloring algorithm to enhance the accessibility for the color vision deficient (or colorblind). Compared to people with normal color vision, peop...

Jia-Bin Huang, Chu-Song Chen, Tzu-Cheng Jen, and S...

posted by Jia-Bin Huang

Read More »

Disclaimer and Copyright Notice

Sciweavers respects the rights of all copyright holders and in this regard, authors are only allowed to share a link to their preprint paper on their own website. Every contribution is associated with a desciptive image. It is the sole responsibility of the authors to ensure that their posted image is not copyright infringing. This service is compliant with IEEE copyright.

Id	Views	Title	Status
1	979	Image Recolorization for the Colorblind
2	145	Compressive confocal microscopy	claim
3	141	Neural network based language models for highly inflective languages	claim
4	124	Rate-distortion optimized bitstream extractor for motion scalability in scalable video coding	claim
5	97	Pitch bends and tonguing articulation in clarinet physical modeling synthesis	claim
6	88	Flatbed scanner identification based on dust and scratches over scanner platen	claim
7	88	Color extended visual cryptography using error diffusion	claim
8	85	Structured variational methods for distributed inference in wireless ad hoc and sensor networks	claim
9	84	Using speech transformation to increase speech intelligibility for the hearing- and speaking-impaired	claim
10	84	Segmentation of malaria parasites in peripheral blood smear images	claim
11	84	Distributed compressive video sensing	claim
12	83	ViBE: A powerful random technique to estimate the background in video sequences	claim
13	82	Registration of multimodal data for estimating the parameters of an articulatory model	claim
14	81	Separable PCA for image classification	claim
15	77	Exploiting statically schedulable regions in dataflow programs	claim
16	73	A watermarking-based method for single-channel audio source separation	claim
17	72	Adaptive dereverberation of speech signals with speaker-position change detection	claim
18	72	Background recovery from video sequences using motion parameters	claim
19	72	Evaluating digital audio authenticity with spectral distances and ENF phase change	claim
20	72	Fixed SINR solutions for the MIMO wiretap channel	claim
21	71	Acoustic fall detection using Gaussian mixture models and GMM supervectors	claim
22	70	Differential feedback of MIMO channel correlation matrices based on geodesic curves	claim
23	69	On the performance of semidefinite relaxation MIMO detectors for QAM constellations	claim
24	69	Blind sparse source separation for unknown number of sources using Gaussian mixture model fitting with Dirichlet prior	claim
25	68	An efficient and robust method for detecting copy-move forgery	claim
26	68	Statistical analysis of amplitude modulation in speech signals using an AM-FM model	claim
27	68	Summarization of large scale social network activity	claim
28	67	DOA estimation of quasi-stationary signals via Khatri-Rao subspace	claim
29	67	CMOS compressed imaging by Random Convolution	claim
30	66	Bandwidth adaptive hardware architecture of K-Means clustering for intelligent video processing	claim
31	66	Microarray classification using block diagonal linear discriminant analysis with embedded feature selection	claim
32	65	Support vector machines and Joint Factor Analysis for speaker verification	claim
33	64	Compression of image patches for local feature extraction	claim
34	64	Range image quality assessment by Structural Similarity	claim
35	63	Multi-view tracking of articulated human motion in silhouette and pose manifolds	claim
36	62	Comparing maximum a posteriori vector quantization and Gaussian mixture models in speaker verification	claim
37	62	Nonparametric curve alignment	claim
38	62	Factor graph based design of an OFDM-IDMA receiver performing joint data detection, channel estimation, and channel length selection	claim
39	62	Joint reconstruction of compressed multi-view images	claim
40	61	Social correlates of turn-taking behavior	claim
41	61	Comparison of scoring methods used in speaker recognition with Joint Factor Analysis	claim
42	61	Rate efficient remote video file synchronization	claim
43	60	Principal component analysis in decomposable Gaussian graphical models	claim
44	60	Jitter compensation in sampling via polynomial least squares estimation	claim
45	59	Complex NMF: A new sparse representation for acoustic signals	claim
46	59	Fast bayesian compressive sensing using Laplace priors	claim
47	59	The data deluge: Challenges and opportunities of unlimited data in statistical signal processing	claim
48	59	TDOA estimation for cyclostationary sources: New correlations-based bounds and estimators	claim
49	59	A fully affine invariant image comparison method	claim
50	59	Interference alignment via alternating minimization	claim
51	58	Predictive power control and multiple-description coding for wireless sensor networks	claim
52	57	Enhancement of reverberant speech using the CELP postfilter	claim
53	57	Secure image retrieval through feature protection	claim
54	57	Thresholded smoothed-l0(SL0) dictionary learning for sparse representations	claim
55	57	Reconstruction of isometrically deformable flat surfaces in 3D from multiple camera images	claim
56	57	Space Kernel Analysis	claim
57	57	Multichannel nonnegative matrix factorization in convolutive mixtures. With application to blind audio source separation	claim
58	56	Identification of neurons participating in cell assemblies	claim
59	56	Small-group learning projects to make signal processing more appealing: From speech processing to OFDMA synchronization	claim
60	56	Denoising scheme for realistic digital photos from unknown sources	claim
61	56	Transmit beamforming for wireless multicasting using channel orthogonalization and local refinement	claim
62	55	Extensions of absolute discounting (Kneser-Ney method)	claim
63	55	An information geometric approach to supervised dimensionality reduction	claim
64	55	A semi-supervised learning approach to online audio background detection	claim
65	55	WEB-derived pronunciations	claim
66	55	Geometrical interpretation and improvements of the Blahut-Arimoto's algorithm	claim
67	55	COSINE - A corpus of multi-party COnversational Speech In Noisy Environments	claim
68	54	Connecting spectral and spring methods for manifold learning	claim
69	54	Energy-efficient graph-based wavelets for distributed coding in Wireless Sensor Networks	claim
70	54	Automatic named identification of speakers using diarization and ASR systems	claim
71	54	Cepstral modulation ratio regression (CMRARE) parameters for audio signal analysis and classification	claim
72	53	Multi-foveation filtering	claim
73	53	Fast acoustic computations using graphics processors	claim
74	53	Incremental predictive and adaptive noise compensation	claim
75	53	What happens when cognitive terminals compete for a relaying node?	claim
76	53	A tempering approach for Itakura-Saito non-negative matrix factorization. With application to music transcription	claim
77	53	Map approach to learning sparse Gaussian Markov networks	claim
78	53	A study of pronunciation verification in a speech therapy application	claim
79	53	Spoken language interpretation: On the use of dynamic Bayesian networks for semantic composition	claim
80	52	Combining VTS model compensation and support vector machines	claim
81	52	A PDE characterization of the intrinsic mode functions	claim
82	52	Transmit/receive beamforming for MIMO radar with colocated antennas	claim
83	52	Automatic evaluation of spoken english fluency	claim
84	52	Language model parameter estimation using user transcriptions	claim
85	52	The gigavision camera	claim
86	52	Humming-based human verification and identification	claim
87	52	Motion effect modeling in multipath configuration using warping based lag-Doppler filtering	claim
88	52	DNA coding using finite-context models and arithmetic coding	claim
89	51	Neighborhood gossip: Concurrent averaging through local interference	claim
90	51	Voice conversion using Artificial Neural Networks	claim
91	51	Audio classification from time-frequency texture	claim
92	51	A global optimization framework for meeting summarization	claim
93	51	Making chroma features more robust to timbre changes	claim
94	51	MIMO decoding based on stochastic reconstruction from multiple projections	claim
95	51	Real-time implementation of robust face detection on mobile platforms	claim
96	50	DCT based multiple hashing technique for robust audio fingerprinting	claim
97	50	Conditional random fields for the prediction of signal peptide cleavage sites	claim
98	50	Dirichlet process mixture models with multiple modalities	claim
99	50	Timing and frequency synchronization for OFDM based cooperative systems	claim
100	50	Vector perturbation precoding for receivers with limited dynamic range	claim
101	50	Combining independent component analysis with geometric information and its application to speech processing	claim
102	50	Exploring functional connectivity in fMRI via clustering	claim
103	49	Time-sensitive behavior dynamics in multimedia fingerprinting social networks	claim
104	49	A game theoretical algorithm for joint power and topology control in distributed WSN	claim
105	49	Motion-based object segmentation using local background sprites	claim
106	49	Video coding using Variable Block-Size Spatially Varying Transforms	claim
107	49	Syntactically-informed models for comma prediction	claim
108	49	Combining discriminative re-ranking and co-training for parsing Mandarin speech transcripts	claim
109	49	Field inversion by consensus and compressed sensing	claim
110	49	Probabilistic matrix tri-factorization	claim
111	49	An auditory-based feature for robust speech recognition	claim
112	49	VLSI for 5000-word continuous speech recognition	claim
113	48	Audiovisual celebrity recognition in unconstrained web videos	claim
114	48	Distributed parameter estimation with selective cooperation	claim
115	48	Contrasting emotion-bearing laughter types in multiparticipant vocal activity detection for meetings	claim
116	48	Cross-lingual speech recognition under runtime resource constraints	claim
117	48	Multiterminal source coding of Bernoulli-Gaussian correlated sources	claim
118	48	A fast feature extraction method	claim
119	48	Multiview video compression and streaming based on predicted viewer position	claim
120	48	Speaker recognition using syllable-based constraints for cepstral frame selection	claim
121	48	A MIMO channel perturbation analysis for robust bit loading	claim
122	48	Sparse LMS for system identification	claim
123	48	GTD-based transceivers for decision feedback and bit loading	claim
124	48	Data-driven online variational filtering in wireless sensor networks	claim
125	47	Log-likelihood ratio clipping in MIMO-BICM systems: Information geometric analysis and impact on system capacity	claim
126	47	Volterra series for analyzing MLP based phoneme posterior estimator	claim
127	47	Incorporating prior knowledge on the digital media creation process into audio classifiers	claim
128	47	Higher dimensional consensus algorithms in sensor networks	claim
129	47	Bispectrum on finite groups	claim
130	47	Quickest change detection in multiple on-off processes	claim
131	47	Design of oversampled DFT modulated filter banks optimized for acoustic echo cancellation	claim
132	47	Optimized opportunistic multicast scheduling (OMS) over heterogeneous cellular networks	claim
133	47	Automatic visual-only language identification: A preliminary study	claim
134	47	Chirp rate estimation of speech based on a time-varying quasi-harmonic model	claim
135	47	A hybrid method for deconvolution of Bernoulli-Gaussian processes	claim
136	47	Unsupervised equalization of Lombard effect for speech recognition in noisy adverse environment	claim
137	47	Compressive wide-band spectrum sensing	claim
138	47	A low complexity Orthogonal Matching Pursuit for sparse signal approximation with shift-invariant dictionaries	claim
139	46	People location and orientation tracking in multiple views	claim
140	46	Periodic event detection and recognition in video	claim
141	46	Robust speech dereverberation based on non-negativity and sparse nature of speech spectrograms	claim
142	46	Benchmarking flexible adaptive time-frequency transforms for underdetermined audio source separation	claim
143	46	RLS-weighted Lasso for adaptive estimation of sparse signals	claim
144	46	The use of acoustically detected filled and silent pauses in spontaneous speech recognition	claim
145	46	Spoken term detection using fast phonetic decoding	claim
146	46	Bayesian discriminative adaptation for speech recognition	claim
147	46	Distance blurring for space-variant image coding	claim
148	46	Postural time-series analysis using Empirical Mode Decomposition and second-order difference plots	claim
149	46	A scale transform based method for rhythmic similarity of music	claim
150	46	Block Jacobi-type methods for non-orthogonal joint diagonalisation	claim
151	46	A variational EM algorithm for learning eigenvoice parameters in mixed signals	claim
152	46	Image spam filtering using Fourier-Mellin invariant features	claim
153	46	Directed Markov Stationary Features for visual classification	claim
154	46	Weighted nonnegative matrix factorization	claim
155	45	One-handed gesture recognition using ultrasonic Doppler sonar	claim
156	45	Robust-SL0 for stable sparse representation in noisy settings	claim
157	45	Instantaneous frequency rate estimation for high-order polynomial-phase signal	claim
158	45	Sparse source separation from orthogonal mixtures	claim
159	45	Compressive sensing for sparsely excited speech signals	claim
160	45	Data pruning-based compression using high order edge-directed interpolation	claim
161	45	A simple, efficient and near optimal algorithm for compressed sensing	claim
162	45	Efficacy of a constantly adaptive language modeling technique for web-scale applications	claim
163	45	Unsupervised pronunciation validation	claim
164	45	The I4U system in NIST 2008 speaker recognition evaluation	claim
165	45	Detection of sparse signals under finite-alphabet constraints	claim
166	45	Modelling the neurovascular habituation effect on fMRI time series	claim
167	45	Using complex-valued ICA to efficiently combine radar polarimetric data for target detection	claim
168	45	Product-HMMs for automatic sign language recognition	claim
169	44	Tree configuration games for distributed stream mining systems	claim
170	44	A modified distortion metric for audio coding	claim
171	44	Voice conversion for various types of body transmitted speech	claim
172	44	Generalized Baum-Welch algorithm for discriminative training on large vocabulary continuous speech recognition system	claim
173	44	Distributed sensing of signals linked by sparse filtering	claim
174	44	Time-varying Opportunistic Protocol for maximizing sensor networks lifetime	claim
175	44	A framework for distributed multimedia stream mining systems using coalition-based foresighted strategies	claim
176	44	On the effect of transmitter IQ imbalance at OFDMA receivers	claim
177	44	Particle filtering for Quantized Innovations	claim
178	44	Audio-assisted trajectory estimation in non-overlapping multi-camera networks	claim
179	44	Analog flat filter design	claim
180	44	Frequency invariant MVDR beamforming without filters and implementation using MIMO radar	claim
181	44	L1 regularized super-resolution from unregistered omnidirectional images	claim
182	44	Comparison of convex combination and affine combination of adaptive filters	claim
183	44	On acoustic surveillance of hazardous situations	claim
184	44	Novel schemes for nonlinear acoustic echo cancellation based on filter combinations	claim
185	44	Rational canonical form of polyphase matrices with applications to designing paraunitary filter banks	claim
186	43	A MIMO-OFDM channel estimation scheme utilizing complementary sequences	claim
187	43	On robustness of coupled adaptive filters	claim
188	43	THE SRI NIST 2008 speaker recognition evaluation system	claim
189	43	Compressive sampling of pulse trains: Spread the spectrum!	claim
190	43	Inpainting with sparse linear combinations of exemplars	claim
191	43	A sum-of-products model for effective coherent modulation filtering	claim
192	43	Genre effects on automatic sentence segmentation of speech: A comparison of broadcast news and broadcast conversations	claim
193	43	Improving multi-lattice alignment based spoken keyword spotting	claim
194	43	Progressive lossless compression of medical images	claim
195	43	Compressive spectral estimation for nonstationary random processes	claim
196	43	A compressive sensing approach to object-based surveillance video coding	claim
197	43	Joint map adaptation of feature transformation and Gaussian Mixture Model for speaker recognition	claim
198	43	Time-space-sequential algorithms for distributed Bayesian state estimation in serial sensor networks	claim
199	43	Distributed adaptive estimation of correlated node-specific signals in a fully connected sensor network	claim
200	43	Statistics for complex random variables revisited	claim
201	43	Minimum variance modulation filter for robust speech recognition	claim
202	43	Subspace tracking of fast time-varying channels in precoded MIMO-OFDM systems	claim
203	43	Color image feature extraction using color index local auto-correlations	claim
204	43	Using fourier-based shape alignment to add geometric prior to snakes	claim
205	42	Fast LCD motion deblurring by decimation and optimization	claim
206	42	Speech emotion recognition via a max-margin framework incorporating a loss function based on the Watson and Tellegen's emotion model	claim
207	42	Filtering web text to match target genres	claim
208	42	Performance bounds of MIMO receivers in the presence of radio frequency interference	claim
209	42	Improved subspace DoA estimation methods with large arrays: The deterministic signals case	claim
210	42	Cheat-proof cooperation strategies for wireless live streaming social networks	claim
211	42	Bayesian Pursuit algorithm for sparse representation	claim
212	42	Sampling signals with finite rate of innovation in the presence of noise	claim
213	42	Document reconstruction using dynamic programming	claim
214	42	Affine invariant features and their application to speech recognition	claim
215	42	Robust discriminative keyword spotting for emotionally colored spontaneous speech using bidirectional LSTM networks	claim
216	42	Learning to maximize signal-to-noise ratio for reverberant speech segregation	claim
217	42	Maximizing global entropy reduction for active learning in speech recognition	claim
218	42	Tampering identification using Empirical Frequency Response	claim
219	42	Dynamic texture models of music	claim
220	42	Blind noise variance estimation for OFDMA signals	claim
221	42	Analyzing Least Squares and Kalman Filtered Compressed Sensing	claim
222	42	Multi-microphone maximum a posteriori fundamental frequency estimation in the cepstral domain	claim
223	42	Control of prosodic focus in corpus-based generation of fundamental frequency contours of Japanese based on the generation process model	claim
224	42	Qualitative analysis of rotational modes within three dimensional empirical mode decomposition	claim
225	42	A new perceptual quality metric for compressed video	claim
226	41	A performance-weighted mixture of LMS filters	claim
227	41	Minimum BER beamforming in the RF domain for OFDM transmissions and linear receivers	claim
228	41	Cultural style based music classification of audio signals	claim
229	41	Data-driven lexicon expansion for Mandarin broadcast news and conversation speech recognition	claim
230	41	A frequency hopping spread spectrum transmission scheme for uncoordinated cognitive radios	claim
231	41	A nonparametric test for stationarity based on local Fourier analysis	claim
232	41	Fusing short term and long term features for improved speaker diarization	claim
233	41	Fingerprint matching based on distance metric learning	claim
234	41	Bounded conditional mean imputation with Gaussian mixture models: A reconstruction approach to partly occluded features	claim
235	41	Real-time dynamic MR image reconstruction using Kalman Filtered Compressed Sensing	claim
236	41	Robust cross-race gene expression analysis	claim
237	41	Strategies for modeling reverberant speech in the feature domain	claim
238	41	Scalable superwideband extension for wideband coding	claim
239	41	Decentralized variational filtering for simultaneous sensor localization and target tracking in binary sensor networks	claim
240	41	Design of robust superdirective beamformers as a convex optimization problem	claim
241	41	Evolution of social P2P networks based on the dynamics of heterogeneous multimedia peers	claim
242	41	Robust Bayesian tracking on Riemannian manifolds via fragments-based representation	claim
243	41	Applying discretized articulatory knowledge to dysarthric speech	claim
244	41	A multiplicative weight perturbation scheme for distributed beamforming in wireless relay networks with 1-bit feedback	claim
245	40	Two microphone based direction of arrival estimation for multiple speech sources using spectral properties of speech	claim
246	40	Applications of complex augmented kernels to wind profile prediction	claim
247	40	A study on multilingual acoustic modeling for large vocabulary ASR	claim
248	40	A wavelet-based quadratic extension method for image deconvolution in the presence of poisson noise	claim
249	40	Phase-based alignment of two signals having partially overlapped spectra	claim
250	40	Query by tapping system based on alignment algorithm	claim
251	40	Geometric distortion signatures for printer identification	claim
252	40	Data hiding in hard-copy text documents robust to print, scan and photocopy operations	claim
253	40	Generalized mutual interdependence analysis	claim
254	40	Law recognition via histogram-based estimation	claim
255	40	Accelerated 3D MRI of vocal tract shaping using compressed sensing and parallel imaging	claim
256	40	Robust word boundary detection in spontaneous speech using acoustic and lexical cues	claim
257	40	Multidimensional signal reconstruction from multichannel acquisition	claim
258	40	Low bitrate audio coding using generalized adaptive gain shape vector quantization across channels	claim
259	40	A new method for speaker adaptation using bilinear model	claim
260	40	Dense disparity estimation in a multi-view distributed video coding system	claim
261	40	Unsupervised speaker adaptation for telephone call transcription	claim
262	40	Transcoding based robust streaming of compressed video	claim
263	40	Outage-based designs for multi-user transceivers	claim
264	40	Missing data recovery via a nonparametric iterative adaptive approach	claim
265	40	Fast noise PSD estimation with low complexity	claim
266	39	High improvement of speaker identification and verification by combining MFCC and phase information	claim
267	39	Part-of-speech histograms for genre classification of text	claim
268	39	Layers and layer interfaces in wireless networks	claim
269	39	Combined image plus depth seam carving for multiview 3D images	claim
270	39	Transceiver design with vector perturbation technique and iterative power loading	claim
271	39	Bayesian sparse image reconstruction for MRFM	claim
272	39	On the use of Bayesian modeling for predicting noise reduction performance	claim
273	39	A generalized family of parameter estimation techniques	claim
274	39	On combinations of CMA equalizers	claim
275	39	Design of a morphological moving object signature and application to human identification	claim
276	39	Exploiting user feedback for language model adaptation in meeting recognition	claim
277	39	Covariate shift adaptation for semi-supervised speaker identification	claim
278	39	Long-time span acoustic activity analysis from far-field sensors in smart homes	claim
279	39	The expected amplitude of overlapping partials of harmonic sounds	claim
280	39	Acoustic compensation methods for body transmitted speech conversion	claim
281	39	Joint estimation of short-term and long-term predictors in speech coders	claim
282	39	Singing voice detection in music tracks using direct voice vibrato detection	claim
283	39	Generating high performance pruned FFT implementations	claim
284	39	An EM algorithm for SCFG in formal syntax-based translation	claim
285	39	A blind speech enhancement algorithm for the suppression of late reverberation and noise	claim
286	39	Audio segmentation for speech recognition using segment features	claim
287	39	On separating glottal source and vocal tract information in telephony speaker verification	claim
288	39	Quadtree structured restoration algorithms for piecewise polynomial images	claim
289	39	Joint optimization of the redundancy of multiple-description coders for multicast	claim
290	39	Instantaneous pose estimation using rotation vectors	claim
291	38	Role of head pose estimation in speech acquisition from distant microphones	claim
292	38	A split quaternion nonlinear adaptive filter	claim
293	38	Robust and fast non asymmetric distributed source coding using turbo codes on the syndrome trellis	claim
294	38	Video event detection and summarization using audio, visual and text saliency	claim
295	38	Progressive lossy-to-lossless coding of arbitrarily-sampled image data using the modified scattered data coding method	claim
296	38	Joint MIMO radar waveform and receiving filter optimization	claim
297	38	Consistent recovery of stimuli encoded with a neural ensemble	claim
298	38	Wrapping snakes for improved lip segmentation	claim
299	38	Psychoacoustically constrained and distortion minimized speech enhancement algorithm	claim
300	38	Rate distribution between model and signal for multiple descriptions	claim
301	38	Multi-modal activity and dominance detection in smart meeting rooms	claim
302	38	Posterior features applied to speech recognition tasks with user-defined vocabulary	claim
303	38	Speech reinforcement based on partial masking effect	claim
304	38	A speech fragment approach to localising multiple speakers in reverberant environments	claim
305	38	Resampling auxiliary data for language model adaptation in machine translation for speech	claim
306	38	Training and adapting MLP features for Arabic speech recognition	claim
307	38	Lattice-based MLLR for speaker recognition	claim
308	38	Functional estimation in Hilbert space for distributed learning in wireless sensor networks	claim
309	38	A joint decoding algorithm for multiple-example-based addition of words to a pronunciation lexicon	claim
310	38	Fusion of fMRI, sMRI, and EEG data using canonical correlation analysis	claim
311	38	Interval consensus: From quantized gossip to voting	claim
312	38	Leveraging multiple query logs to improve language models for spoken query recognition	claim
313	38	Rate-constrained distributed distance testing and its applications	claim
314	38	Oscillator Phase Noise compensation using Kalman tracking	claim
315	38	Multi-channel audio segmentation for continuous observation and archival of large spaces	claim
316	38	SkellamShrink: Poisson intensity estimation for vector-valued data	claim
317	38	Multi-modal speaker diarization of real-world meetings using compressed-domain video features	claim
318	38	Using a pitch-synchronous residual codebook for hybrid HMM/frame selection speech synthesis	claim
319	38	Mobile media search	claim
320	37	Sparse Decomposition over non-full-rank dictionaries	claim
321	37	Using collective information in semi-supervised learning for speech recognition	claim
322	37	Annotating images by harnessing worldwide user-tagged photos	claim
323	37	Mutual information based channel selection for speaker diarization of meetings data	claim
324	37	Optimal parameter estimation for model-based quantization	claim
325	37	Sparse decomposition of mixed audio signals by basis pursuit with autoregressive models	claim
326	37	OFDM Turbo DeCodulation with exit optimized bit loading and signal constellations	claim
327	37	Music emotion ranking	claim
328	37	Extended VTS for noise-robust speech recognition	claim
329	37	Hierarchical fusion of color and depth information at partition level by cooperative region merging	claim
330	37	A differential geometric approach to discrete-coefficient filter design	claim
331	37	Gaussian Backend design for open-set language detection	claim
332	37	Improved clustered hierarchical tandem system with bottom-up processing	claim
333	37	Distributed subgradient projection algorithm for convex optimization	claim
334	37	A quality prediction model for JPEG2000-based color images	claim
335	37	Spoken dialog strategy based on understanding graph search	claim
336	37	Restoring punctuation and capitalization in transcribed speech	claim
337	37	From rule-based to statistical grammars: Continuous improvement of large-scale spoken dialog systems	claim
338	37	Relative pitch estimation of multiple instruments	claim
339	37	Detecting bandlimited audio in broadcast television shows	claim
340	37	Robust joint localization and time synchronization in wireless sensor networks with bounded anchor uncertainties	claim
341	37	Automatic musical meter detection	claim
342	37	A mixed time-scale algorithm for distributed parameter estimation : Nonlinear observation models and imperfect communication	claim
343	37	Target detection using incremental learning on single-trial evoked response	claim
344	37	Using phase linearity in frequency-domain ICA to tackle the permutation problem	claim
345	37	Dictionary learning for the sparse modelling of atrial fibrillation in ECG signals	claim
346	37	Using filter banks to enhance images for fluid lens cameras based on color correlation	claim
347	37	Adaptive predistortion of nonlinear Volterra systems using Spectral Magnitude Matching	claim
348	37	Complete characterization of perfectly secure stego-systems with mutually independent embedding operation	claim
349	37	Applying improved spectral modeling for High Quality voice conversion	claim
350	36	Game theory for precoding in a multi-user system: Bargaining for overall benefits	claim
351	36	Voronoi cell shaping for feature selection with discrete HMMs	claim
352	36	The effect of formant trajectories and phoneme durations on vowel intelligibility	claim
353	36	Detecting real life anger	claim
354	36	Separation of a subspace-sparse signal: Algorithms and conditions	claim
355	36	Comparison of MPEG-7 descriptors for long term selection of reference frames	claim
356	36	Multi-level non-rigid image registration using graph-cuts	claim
357	36	Polyphonic musical instrument recognition based on a dynamic model of the spectral envelope	claim
358	36	Spoken language understanding from unaligned data using discriminative classification models	claim
359	36	Modelling the prepausal lengthening effect for speech recognition: a dynamic Bayesian network approach	claim
360	36	Using the scaling ambiguity for filter shortening in convolutive blind source separation	claim
361	36	Underdetermined audio source separation from anechoic mixtures with long time delay	claim
362	36	Adaptive quickest change detection with unknown parameter	claim
363	36	Emotion recognition from speech: Putting ASR in the loop	claim
364	36	Multi-flow attack resistant watermarks for network flows	claim
365	36	Graphical Models: Statistical inference vs. determination	claim
366	36	Multichannel speech enhancement using convolutive transfer function approximation in reverberant environments	claim
367	36	Low-complexity frequency-domain turbo equalization for single-carrier transmissions over doubly-selective channels	claim
368	36	Jointly optimized mode decisions in redundant video streaming	claim
369	36	Wildfire detection using LMS based active learning	claim
370	36	Delay-based overlay construction in P2P video broadcast	claim
371	36	Automatic pronunciation verification of english letter-names for early literacy assessment of preliterate children	claim
372	36	An efficient first-order method for l1 compression of images	claim
373	36	Ensembles of landmark multidimensional scalings	claim
374	36	Narrowband interference mitigation in BICM OFDM systems	claim
375	36	Voice convergin: Speaker de-identification by voice transformation	claim
376	35	Anti-collusion fingerprinting with scalar Costa scheme (SCS) and colluder weight recovery	claim
377	35	Convex analysis based minimum-volume enclosing simplex algorithm for hyperspectral unmixing	claim
378	35	Dense error correction via l1-minimization	claim
379	35	Improving acoustic speaker verification with visual Body-Language features	claim
380	35	Room impulse response shortening with infinity-norm optimization	claim
381	35	Posterior-based confidence measures for spoken term detection	claim
382	35	A Bayesian approach to HMM-based speech synthesis	claim
383	35	Cross-layer optimization of wireless fading ad-hoc networks	claim
384	35	Perceptual quality based packet dropping for generalized video GOP structures	claim
385	35	Language model transformation applied to lightly supervised training of acoustic model for congress meetings	claim
386	35	Optimizing segment label boundaries for statistical speech synthesis	claim
387	35	Formant-based technique for automatic filled-pause detection in spontaneous spoken english	claim
388	35	Modal expansion of HRTFs: Continuous representation in frequency-range-angle	claim
389	35	Robust modeling of musical chord sequences using probabilistic N-grams	claim
390	35	Real-time PDA-based recursive Fourier transform Implementation for cochlear implant applications	claim
391	35	A Bayesian NETWORKS approach for dialog modeling: The fusion BN	claim
392	35	A multistage approach for blind separation of convolutive speech mixtures	claim
393	35	Revisiting graphemes with increasing amounts of data	claim
394	35	Independent component analysis for noisy speech recognition	claim
395	35	Adaptive coding of images via multiresolution ICA	claim
396	35	A fast asymptotically efficient algorithm for blind separation of a linear mixture of block-wise stationary autoregressive processes	claim
397	35	Sparse imputation for noise robust speech recognition using soft masks	claim
398	35	Distributions of 3D DCT coefficients for video	claim
399	34	Sparse boosting	claim
400	34	Estimating correspondence between multiple cameras using joint invariants	claim
401	34	Combining frontend-based memory with MFCC features for Bandwidth Extension of narrowband speech	claim
402	34	On the role of localization cues in binaural segregation of reverberant speech	claim
403	34	Polyphase interpretation of empirical image interpolation	claim
404	34	On a tradeoff between dereverberation and noise reduction using the MVDR beamformer	claim
405	34	High-rate distributed multi-source cooperation using complex field coding	claim
406	34	Low-delay scheduling for Grassmannian beamforming with a SINR constraint	claim
407	34	Co-adaptation: Adaptive co-training for semi-supervised learning	claim
408	34	Adaptive distributed transforms for irregularly sampled Wireless Sensor Networks	claim
409	34	Active learning for semi-supervised multi-task learning	claim
410	34	High-level feature extraction using SVM with walk-based graph kernel	claim
411	34	Improved lattice-based spoken document retrieval by directly learning from the evaluation measures	claim
412	34	High resolution audio synchronization using chroma onset features	claim
413	34	Distributed peer-to-peer beamforming for multiuser relay networks	claim
414	34	Modeling of contours in wavelet domain for generalized lifting image compression	claim
415	34	Phoneme cluster based state mapping for text-independent voice conversion	claim
416	34	Automatic voice assignment tool for Instant Casting movie System	claim
417	34	Visual grouping by neural oscillators	claim
418	34	Intensity vector direction exploitation for exhaustive blind source separation of convolutive mixtures	claim
419	34	Semi-definite programming approach to sensor network node localization with anchor position uncertainty	claim
420	34	Maximum-likelihood estimation of autoregressive models with conditional independence constraints	claim
421	34	Q-SIFT: Efficient feature descriptors for distributed camera calibration	claim
422	34	Incorporating monolingual corpora into bilingual latent semantic analysis for crosslingual LM adaptation	claim
423	34	Signal processing challenges for future wireless communications	claim
424	33	Blind system identification for speech dereverberation with Forced Spectral Diversity	claim
425	33	Sparse decomposition of two dimensional signals	claim
426	33	Scalable distributed source coding	claim
427	33	Training-based Bayesian MIMO channel and channel norm estimation	claim
428	33	Estimating multiple transmitter locations from power measurements at multiple receivers	claim
429	33	Blind Monte Carlo detection-estimation method for optical coherence tomography	claim
430	33	A low-complexity spectro-temporal based perceptual model	claim
431	33	Testing fractal connectivity in multivariate long memory processes	claim
432	33	Enhanced error resilience of video communications for burst losses using an extended ROPE algorithm	claim
433	33	A reduced-reference video structural similarity metric based on no-reference estimation of channel-induced distortion	claim
434	33	A PARAFAC-based technique for detection and localization of multiple targets in a MIMO radar system	claim
435	33	Generic invertibility of multidimensional FIR multirate systems and filter banks	claim
436	33	Phonological features in discriminative classification of dysarthric speech	claim
437	33	On perfect channel identifiability of semiblind ML detection of orthogonal space-time block coded OFDM	claim
438	33	Monaural voiced speech segregation based on elaborate harmonic grouping strategy	claim
439	33	Trajectory training considering global variance for HMM-based speech synthesis	claim
440	33	The speed of greed: Characterizing myopic gossip through network voracity	claim
441	33	Beacon-assisted spectrum access with cooperative cognitive transmitter and receiver	claim
442	33	Distributed sampling and reconstruction of non-bandlimited fields in sensor networks based on shift-invariant spaces	claim
443	32	GridNews: A distributed automatic Greek broadcast transcription system	claim
444	32	Interpolating hidden Markov model and its application to automatic instrument recognition	claim
445	32	On the phonetic information in ultrasonic microphone signals	claim
446	32	Robust video fingerprinting based on visual attention regions	claim
447	32	Combining mixture weight pruning and quantization for small-footprint speech recognition	claim
448	32	Discriminative training of hierarchical acoustic models for large vocabulary continuous speech recognition	claim
449	32	A new stochastic estimator for tremor frequency tracking	claim
450	32	An analytical approach to sound field reproduction with a movable sweet spot using circular distributions of loudspeakers	claim
451	32	Exploiting T-junctions for depth segregation in single images	claim
452	32	On the error exponents for detecting randomly sampled noisy diffusion processes	claim
453	32	A criterion for the enhancement of time-frequency masks in missing data recognition	claim
454	32	Fast mean square convergence of consensus algorithms in WSNs with random topologies	claim
455	32	Filter-and-forward distributed beamforming for relay networks in frequency selective fading channels	claim
456	32	Unsupervised acoustic and language model training with small amounts of labelled data	claim
457	32	Cache-based integer motion/disparity estimation for quad-HD H.264/AVC and HD multiview video coding	claim
458	31	Resource usage prediction for groups of dynamic image-processing tasks using Markov modeling	claim
459	31	Multi-layer ratio Semi-Definite Classifiers	claim
460	31	Parsimonious dictionary learning	claim
461	31	Bandwidth-efficient cache-based motion compensation architecture with DRAM-friendly data access control	claim
462	31	Modeling and tracking of faces in real-life illumination conditions	claim
463	31	Spatio-temporal adaptive detector in non-homogeneous and low-rank clutter	claim
464	31	On the recovery of nonnegative sparse vectors from sparse measurements inspired by expanders	claim
465	31	Speech rhythm guided syllable nuclei detection	claim
466	31	Impact of novel sources on content-based image and video retrieval	claim
467	31	Fast belief propagation process element for high-quality stereo estimation	claim
468	30	A regularized kernel-based approach to unsupervised audio segmentation	claim
469	30	Plane-wave decomposition of a sound scene using a cylindrical microphone array	claim
470	30	Minimax design of adjustable FIR filters using 2D polynomial methods	claim
471	30	Single and multichannel sampling of bilevel polygons using exponential splines	claim
472	30	Shrinkage estimation of high dimensional covariance matrices	claim
473	30	Experimenting with a global decision tree for state clustering in automatic speech recognition systems	claim
474	30	Reducing F0 Frame Error of F0 tracking algorithms under noisy conditions with an unvoiced/voiced classification frontend	claim
475	30	On ICA of improper and noncircular sources	claim
476	30	The effectiveness of histogram equalization on environmental model adaptation	claim
477	29	A flat direct model for speech recognition	claim
478	29	Ranging energy optimization for robust sensor positioning	claim
479	29	Risk-distortion analysis for video collusion attack	claim
480	29	Self-optimizing scheme for active noise and vibration control	claim
481	28	A game-theoretic framework for multi-user multimedia rate allocation	claim
482	28	A complex cross-spectral distribution model using Normal Variance Mean Mixtures	claim
483	28	Incorporating spectral subtraction and noise type for unvoiced speech segregation	claim
484	28	LSH banding for large-scale retrieval with memory and recall constraints	claim
485	28	Voice search of structured media data	claim
486	28	You are fired! Nonverbal role analysis in competitive meetings	claim
487	28	Spatial super-resolution of a diffusion field by temporal oversampling in sensor networks	claim
488	28	Probablistic modelling of F0 in unvoiced regions in HMM based speech synthesis	claim
489	28	Facial age estimation by multilinear subspace analysis	claim
490	28	The fractional Hilbert transform and dual-tree Gabor-like wavelet analysis	claim
491	28	Quality control of automatic labelling using HMM-based synthesis	claim
492	28	A differential motion estimation method for image interpolation in distributed video coding	claim
493	28	Candidate proposal for ITU-T super-wideband speech and audio coding	claim
494	27	Modeling instantaneous intonation for speaker identification using the fundamental frequency variation spectrum	claim
495	27	Generalization of specialized on-the-fly composition	claim
496	27	One-pass multi-layer rate-distortion optimization for quality scalable video coding	claim
497	26	View-invariant tensor null-space representation for multiple motion trajectory retrieval and classification	claim
498	26	Visual saliency with side information	claim
499	24	Expectation-maximization algorithm for multi-pitch estimation and separation of overlapping harmonic spectra	claim
500	24	An iterative approach to monaural musical mixture de-soloing	claim

Sciweavers

Explore & Download

Productivity Tools

Document Tools

Image Tools

Sciweavers