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ICASSP 2009 Speech Acoustics

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ICASSP
2009
IEEE

1733views Signal Processing» more ICASSP 2009»

Image Recolorization for the Colorblind

14 years 11 months ago

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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

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Id	Views	Title	Status

1	1733	Image Recolorization for the Colorblind
2	496	Neural network based language models for highly inflective languages	claim
3	439	Reconstruction of isometrically deformable flat surfaces in 3D from multiple camera images	claim
4	357	Color extended visual cryptography using error diffusion	claim
5	329	The data deluge: Challenges and opportunities of unlimited data in statistical signal processing	claim
6	319	Compressive confocal microscopy	claim
7	291	Probabilistic matrix tri-factorization	claim
8	287	Rate-distortion optimized bitstream extractor for motion scalability in scalable video coding	claim
9	260	Blind sparse source separation for unknown number of sources using Gaussian mixture model fitting with Dirichlet prior	claim
10	255	Complex NMF: A new sparse representation for acoustic signals	claim
11	243	Principal component analysis in decomposable Gaussian graphical models	claim
12	241	Using speech transformation to increase speech intelligibility for the hearing- and speaking-impaired	claim
13	240	A study of pronunciation verification in a speech therapy application	claim
14	238	Distributed compressive video sensing	claim
15	237	Adaptive dereverberation of speech signals with speaker-position change detection	claim
16	232	Acoustic fall detection using Gaussian mixture models and GMM supervectors	claim
17	226	DCT based multiple hashing technique for robust audio fingerprinting	claim
18	223	CMOS compressed imaging by Random Convolution	claim
19	219	Multi-view tracking of articulated human motion in silhouette and pose manifolds	claim
20	218	Multichannel nonnegative matrix factorization in convolutive mixtures. With application to blind audio source separation	claim
21	218	Robust Bayesian tracking on Riemannian manifolds via fragments-based representation	claim
22	215	Decentralized variational filtering for simultaneous sensor localization and target tracking in binary sensor networks	claim
23	215	Fast acoustic computations using graphics processors	claim
24	214	Microarray classification using block diagonal linear discriminant analysis with embedded feature selection	claim
25	212	One-handed gesture recognition using ultrasonic Doppler sonar	claim
26	212	Range image quality assessment by Structural Similarity	claim
27	211	Segmentation of malaria parasites in peripheral blood smear images	claim
28	211	Compression of image patches for local feature extraction	claim
29	210	Color image feature extraction using color index local auto-correlations	claim
30	209	Distributed adaptive estimation of correlated node-specific signals in a fully connected sensor network	claim
31	209	Combining independent component analysis with geometric information and its application to speech processing	claim
32	207	Modeling and tracking of faces in real-life illumination conditions
33	207	Minimum BER beamforming in the RF domain for OFDM transmissions and linear receivers	claim
34	206	Contrasting emotion-bearing laughter types in multiparticipant vocal activity detection for meetings	claim
35	206	Joint reconstruction of compressed multi-view images	claim
36	205	Hierarchical fusion of color and depth information at partition level by cooperative region merging	claim
37	204	Pitch bends and tonguing articulation in clarinet physical modeling synthesis	claim
38	203	Structured variational methods for distributed inference in wireless ad hoc and sensor networks	claim
39	202	Background recovery from video sequences using motion parameters	claim
40	200	Registration of multimodal data for estimating the parameters of an articulatory model	claim
41	200	Fusion of fMRI, sMRI, and EEG data using canonical correlation analysis	claim
42	198	Compressive wide-band spectrum sensing	claim
43	197	Audio classification from time-frequency texture	claim
44	197	Fixed SINR solutions for the MIMO wiretap channel	claim
45	195	Interference alignment via alternating minimization	claim
46	193	Differential feedback of MIMO channel correlation matrices based on geodesic curves	claim
47	192	Comparison of scoring methods used in speaker recognition with Joint Factor Analysis	claim
48	192	Timing and frequency synchronization for OFDM based cooperative systems	claim
49	191	Comparing maximum a posteriori vector quantization and Gaussian mixture models in speaker verification	claim
50	191	Separation of a subspace-sparse signal: Algorithms and conditions	claim
51	190	ViBE: A powerful random technique to estimate the background in video sequences	claim
52	188	Combined image plus depth seam carving for multiview 3D images	claim
53	188	Using the scaling ambiguity for filter shortening in convolutive blind source separation	claim
54	188	Support vector machines and Joint Factor Analysis for speaker verification	claim
55	187	RLS-weighted Lasso for adaptive estimation of sparse signals	claim
56	187	Summarization of large scale social network activity	claim
57	187	Weighted nonnegative matrix factorization	claim
58	186	Adaptive coding of images via multiresolution ICA	claim
59	185	A differential geometric approach to discrete-coefficient filter design	claim
60	184	Multiterminal source coding of Bernoulli-Gaussian correlated sources	claim
61	183	Bandwidth adaptive hardware architecture of K-Means clustering for intelligent video processing	claim
62	183	A semi-supervised learning approach to online audio background detection	claim
63	183	Automatic named identification of speakers using diarization and ASR systems	claim
64	183	Log-likelihood ratio clipping in MIMO-BICM systems: Information geometric analysis and impact on system capacity	claim
65	182	Field inversion by consensus and compressed sensing	claim
66	182	Genre effects on automatic sentence segmentation of speech: A comparison of broadcast news and broadcast conversations	claim
67	182	Audio segmentation for speech recognition using segment features	claim
68	181	Data-driven online variational filtering in wireless sensor networks	claim
69	181	Conditional random fields for the prediction of signal peptide cleavage sites	claim
70	180	Delay-based overlay construction in P2P video broadcast	claim
71	180	Multi-modal activity and dominance detection in smart meeting rooms	claim
72	180	Time-varying Opportunistic Protocol for maximizing sensor networks lifetime	claim
73	180	Accelerated 3D MRI of vocal tract shaping using compressed sensing and parallel imaging	claim
74	180	An efficient and robust method for detecting copy-move forgery	claim
75	180	Data pruning-based compression using high order edge-directed interpolation	claim
76	179	A compressive sensing approach to object-based surveillance video coding	claim
77	179	The speed of greed: Characterizing myopic gossip through network voracity	claim
78	179	Audiovisual celebrity recognition in unconstrained web videos	claim
79	178	Factor graph based design of an OFDM-IDMA receiver performing joint data detection, channel estimation, and channel length selection	claim
80	178	Independent component analysis for noisy speech recognition	claim
81	178	A hybrid method for deconvolution of Bernoulli-Gaussian processes	claim
82	178	A watermarking-based method for single-channel audio source separation	claim
83	178	Fast bayesian compressive sensing using Laplace priors	claim
84	177	An auditory-based feature for robust speech recognition	claim
85	177	DOA estimation of quasi-stationary signals via Khatri-Rao subspace	claim
86	177	Geometrical interpretation and improvements of the Blahut-Arimoto's algorithm	claim
87	176	Map approach to learning sparse Gaussian Markov networks	claim
88	176	Syntactically-informed models for comma prediction	claim
89	176	Dirichlet process mixture models with multiple modalities	claim
90	175	Denoising scheme for realistic digital photos from unknown sources	claim
91	175	Enhancement of reverberant speech using the CELP postfilter	claim
92	175	Active learning for semi-supervised multi-task learning	claim
93	175	TDOA estimation for cyclostationary sources: New correlations-based bounds and estimators	claim
94	175	Real-time implementation of robust face detection on mobile platforms	claim
95	175	Automatic evaluation of spoken english fluency	claim
96	174	Transmit/receive beamforming for MIMO radar with colocated antennas	claim
97	174	Flatbed scanner identification based on dust and scratches over scanner platen	claim
98	174	Resampling auxiliary data for language model adaptation in machine translation for speech	claim
99	174	Exploiting statically schedulable regions in dataflow programs	claim
100	174	Compressive sensing for sparsely excited speech signals	claim
101	174	Statistical analysis of amplitude modulation in speech signals using an AM-FM model	claim
102	172	Data-driven lexicon expansion for Mandarin broadcast news and conversation speech recognition	claim
103	172	Predictive power control and multiple-description coding for wireless sensor networks	claim
104	172	Robust joint localization and time synchronization in wireless sensor networks with bounded anchor uncertainties	claim
105	172	Image spam filtering using Fourier-Mellin invariant features	claim
106	171	Distributions of 3D DCT coefficients for video	claim
107	170	Robust-SL0 for stable sparse representation in noisy settings	claim
108	170	Compressive spectral estimation for nonstationary random processes	claim
109	169	Annotating images by harnessing worldwide user-tagged photos	claim
110	169	A PDE characterization of the intrinsic mode functions	claim
111	169	Comparison of MPEG-7 descriptors for long term selection of reference frames	claim
112	169	Bayesian Pursuit algorithm for sparse representation	claim
113	169	Chirp rate estimation of speech based on a time-varying quasi-harmonic model	claim
114	169	Social correlates of turn-taking behavior	claim
115	168	Particle filtering for Quantized Innovations	claim
116	168	Multi-microphone maximum a posteriori fundamental frequency estimation in the cepstral domain	claim
117	167	A low complexity Orthogonal Matching Pursuit for sparse signal approximation with shift-invariant dictionaries	claim
118	167	On the performance of semidefinite relaxation MIMO detectors for QAM constellations	claim
119	167	On acoustic surveillance of hazardous situations	claim
120	167	Adaptive predistortion of nonlinear Volterra systems using Spectral Magnitude Matching	claim
121	166	Jitter compensation in sampling via polynomial least squares estimation	claim
122	166	COSINE - A corpus of multi-party COnversational Speech In Noisy Environments	claim
123	166	A MIMO-OFDM channel estimation scheme utilizing complementary sequences	claim
124	165	Multi-level non-rigid image registration using graph-cuts	claim
125	165	Distributed subgradient projection algorithm for convex optimization	claim
126	165	Exploring functional connectivity in fMRI via clustering	claim
127	165	Improved subspace DoA estimation methods with large arrays: The deterministic signals case	claim
128	164	A fast feature extraction method	claim
129	164	Phonological features in discriminative classification of dysarthric speech	claim
130	163	Motion-based object segmentation using local background sprites	claim
131	163	MIMO decoding based on stochastic reconstruction from multiple projections	claim
132	163	Fast noise PSD estimation with low complexity	claim
133	163	An information geometric approach to supervised dimensionality reduction	claim
134	163	A reduced-reference video structural similarity metric based on no-reference estimation of channel-induced distortion	claim
135	163	Spoken language interpretation: On the use of dynamic Bayesian networks for semantic composition	claim
136	162	Filter-and-forward distributed beamforming for relay networks in frequency selective fading channels	claim
137	162	DNA coding using finite-context models and arithmetic coding	claim
138	162	A game theoretical algorithm for joint power and topology control in distributed WSN	claim
139	161	Distributed parameter estimation with selective cooperation	claim
140	161	Cheat-proof cooperation strategies for wireless live streaming social networks	claim
141	161	Space Kernel Analysis	claim
142	161	Time-sensitive behavior dynamics in multimedia fingerprinting social networks	claim
143	161	Missing data recovery via a nonparametric iterative adaptive approach	claim
144	161	Distributed sampling and reconstruction of non-bandlimited fields in sensor networks based on shift-invariant spaces	claim
145	160	The I4U system in NIST 2008 speaker recognition evaluation	claim
146	160	People location and orientation tracking in multiple views	claim
147	160	A modified distortion metric for audio coding	claim
148	160	Adaptive quickest change detection with unknown parameter	claim
149	160	Automatic musical meter detection	claim
150	158	A simple, efficient and near optimal algorithm for compressed sensing	claim
151	158	Scalable distributed source coding	claim
152	158	The expected amplitude of overlapping partials of harmonic sounds	claim
153	157	Analyzing Least Squares and Kalman Filtered Compressed Sensing	claim
154	157	Mutual information based channel selection for speaker diarization of meetings data	claim
155	157	A criterion for the enhancement of time-frequency masks in missing data recognition	claim
156	157	A fast asymptotically efficient algorithm for blind separation of a linear mixture of block-wise stationary autoregressive processes	claim
157	157	Cultural style based music classification of audio signals	claim
158	157	On combinations of CMA equalizers	claim
159	157	Fast mean square convergence of consensus algorithms in WSNs with random topologies	claim
160	156	Automatic visual-only language identification: A preliminary study	claim
161	156	Robust speech dereverberation based on non-negativity and sparse nature of speech spectrograms	claim
162	156	Fingerprint matching based on distance metric learning	claim
163	156	The use of acoustically detected filled and silent pauses in spontaneous speech recognition	claim
164	156	A blind speech enhancement algorithm for the suppression of late reverberation and noise	claim
165	156	Secure image retrieval through feature protection	claim
166	156	Singing voice detection in music tracks using direct voice vibrato detection	claim
167	156	Volterra series for analyzing MLP based phoneme posterior estimator	claim
168	155	Quadtree structured restoration algorithms for piecewise polynomial images	claim
169	155	Connecting spectral and spring methods for manifold learning	claim
170	155	Benchmarking flexible adaptive time-frequency transforms for underdetermined audio source separation	claim
171	155	Humming-based human verification and identification	claim
172	155	Filtering web text to match target genres	claim
173	155	Separable PCA for image classification	claim
174	155	Unsupervised speaker adaptation for telephone call transcription	claim
175	154	Beacon-assisted spectrum access with cooperative cognitive transmitter and receiver	claim
176	154	Covariate shift adaptation for semi-supervised speaker identification	claim
177	154	Making chroma features more robust to timbre changes	claim
178	154	Minimum variance modulation filter for robust speech recognition	claim
179	154	Product-HMMs for automatic sign language recognition	claim
180	154	Two microphone based direction of arrival estimation for multiple speech sources using spectral properties of speech	claim
181	154	Rate efficient remote video file synchronization	claim
182	154	Cepstral modulation ratio regression (CMRARE) parameters for audio signal analysis and classification	claim
183	153	Vector perturbation precoding for receivers with limited dynamic range	claim
184	153	Unsupervised pronunciation validation	claim
185	153	Design of oversampled DFT modulated filter banks optimized for acoustic echo cancellation	claim
186	153	Semi-definite programming approach to sensor network node localization with anchor position uncertainty	claim
187	153	Optimized opportunistic multicast scheduling (OMS) over heterogeneous cellular networks	claim
188	153	A tempering approach for Itakura-Saito non-negative matrix factorization. With application to music transcription	claim
189	153	A PARAFAC-based technique for detection and localization of multiple targets in a MIMO radar system	claim
190	153	Applying discretized articulatory knowledge to dysarthric speech	claim
191	152	Control of prosodic focus in corpus-based generation of fundamental frequency contours of Japanese based on the generation process model	claim
192	152	Instantaneous pose estimation using rotation vectors	claim
193	152	A regularized kernel-based approach to unsupervised audio segmentation	claim
194	152	Generalized Baum-Welch algorithm for discriminative training on large vocabulary continuous speech recognition system	claim
195	152	Identification of neurons participating in cell assemblies	claim
196	151	Evaluating digital audio authenticity with spectral distances and ENF phase change	claim
197	151	Voice conversion using Artificial Neural Networks	claim
198	151	Robust and fast non asymmetric distributed source coding using turbo codes on the syndrome trellis	claim
199	151	Nonparametric curve alignment	claim
200	151	Improved lattice-based spoken document retrieval by directly learning from the evaluation measures	claim
201	150	Maximizing global entropy reduction for active learning in speech recognition	claim
202	150	Combining VTS model compensation and support vector machines	claim
203	150	Voronoi cell shaping for feature selection with discrete HMMs	claim
204	150	Language model parameter estimation using user transcriptions	claim
205	150	Bispectrum on finite groups	claim
206	150	A framework for distributed multimedia stream mining systems using coalition-based foresighted strategies	claim
207	150	Affine invariant features and their application to speech recognition	claim
208	150	Postural time-series analysis using Empirical Mode Decomposition and second-order difference plots	claim
209	150	Candidate proposal for ITU-T super-wideband speech and audio coding	claim
210	150	Incorporating prior knowledge on the digital media creation process into audio classifiers	claim
211	150	Dense disparity estimation in a multi-view distributed video coding system	claim
212	149	Novel schemes for nonlinear acoustic echo cancellation based on filter combinations	claim
213	149	Energy-efficient graph-based wavelets for distributed coding in Wireless Sensor Networks	claim
214	149	Adaptive distributed transforms for irregularly sampled Wireless Sensor Networks	claim
215	148	A split quaternion nonlinear adaptive filter	claim
216	147	Multiview video compression and streaming based on predicted viewer position	claim
217	147	Statistics for complex random variables revisited	claim
218	147	Sparse imputation for noise robust speech recognition using soft masks	claim
219	147	Voice convergin: Speaker de-identification by voice transformation	claim
220	147	Thresholded smoothed-l0(SL0) dictionary learning for sparse representations	claim
221	147	Compressive sampling of pulse trains: Spread the spectrum!	claim
222	147	On the effect of transmitter IQ imbalance at OFDMA receivers	claim
223	147	Distance blurring for space-variant image coding	claim
224	147	Evolution of social P2P networks based on the dynamics of heterogeneous multimedia peers	claim
225	146	Time-space-sequential algorithms for distributed Bayesian state estimation in serial sensor networks	claim
226	146	Speech emotion recognition via a max-margin framework incorporating a loss function based on the Watson and Tellegen's emotion model	claim
227	146	Lattice-based MLLR for speaker recognition	claim
228	146	A fully affine invariant image comparison method	claim
229	146	Language model transformation applied to lightly supervised training of acoustic model for congress meetings	claim
230	146	Sampling signals with finite rate of innovation in the presence of noise	claim
231	146	Detection of sparse signals under finite-alphabet constraints	claim
232	146	Intensity vector direction exploitation for exhaustive blind source separation of convolutive mixtures	claim
233	145	Automatic pronunciation verification of english letter-names for early literacy assessment of preliterate children	claim
234	145	A global optimization framework for meeting summarization	claim
235	145	Wildfire detection using LMS based active learning	claim
236	145	Interpolating hidden Markov model and its application to automatic instrument recognition	claim
237	145	Incremental predictive and adaptive noise compensation	claim
238	145	Generating high performance pruned FFT implementations	claim
239	144	Geometric distortion signatures for printer identification	claim
240	144	Functional estimation in Hilbert space for distributed learning in wireless sensor networks	claim
241	144	Leveraging multiple query logs to improve language models for spoken query recognition	claim
242	144	A mixed time-scale algorithm for distributed parameter estimation : Nonlinear observation models and imperfect communication	claim
243	144	Spoken language understanding from unaligned data using discriminative classification models	claim
244	143	Transmit beamforming for wireless multicasting using channel orthogonalization and local refinement	claim
245	143	Formant-based technique for automatic filled-pause detection in spontaneous spoken english	claim
246	143	Joint map adaptation of feature transformation and Gaussian Mixture Model for speaker recognition	claim
247	143	A complex cross-spectral distribution model using Normal Variance Mean Mixtures	claim
248	143	Strategies for modeling reverberant speech in the feature domain	claim
249	143	High resolution audio synchronization using chroma onset features	claim
250	143	Rational canonical form of polyphase matrices with applications to designing paraunitary filter banks	claim
251	142	A Bayesian NETWORKS approach for dialog modeling: The fusion BN	claim
252	142	Bayesian discriminative adaptation for speech recognition	claim
253	142	Modeling instantaneous intonation for speaker identification using the fundamental frequency variation spectrum	claim
254	142	Reducing F0 Frame Error of F0 tracking algorithms under noisy conditions with an unvoiced/voiced classification frontend	claim
255	142	OFDM Turbo DeCodulation with exit optimized bit loading and signal constellations	claim
256	142	A new method for speaker adaptation using bilinear model	claim
257	142	VLSI for 5000-word continuous speech recognition	claim
258	141	Analog flat filter design	claim
259	141	Probablistic modelling of F0 in unvoiced regions in HMM based speech synthesis	claim
260	141	The fractional Hilbert transform and dual-tree Gabor-like wavelet analysis	claim
261	141	Sparse LMS for system identification	claim
262	141	A flat direct model for speech recognition	claim
263	141	A speech fragment approach to localising multiple speakers in reverberant environments	claim
264	141	Blind Monte Carlo detection-estimation method for optical coherence tomography	claim
265	141	Inpainting with sparse linear combinations of exemplars	claim
266	141	Law recognition via histogram-based estimation	claim
267	141	Robust cross-race gene expression analysis	claim
268	141	Interval consensus: From quantized gossip to voting	claim
269	141	Speech reinforcement based on partial masking effect	claim
270	141	Blind noise variance estimation for OFDMA signals	claim
271	140	THE SRI NIST 2008 speaker recognition evaluation system	claim
272	140	Oscillator Phase Noise compensation using Kalman tracking	claim
273	140	Dynamic texture models of music	claim
274	140	What happens when cognitive terminals compete for a relaying node?	claim
275	140	Multidimensional signal reconstruction from multichannel acquisition	claim
276	140	Comparison of convex combination and affine combination of adaptive filters	claim
277	139	Low-complexity frequency-domain turbo equalization for single-carrier transmissions over doubly-selective channels	claim
278	139	Speaker recognition using syllable-based constraints for cepstral frame selection	claim
279	139	Multi-modal speaker diarization of real-world meetings using compressed-domain video features	claim
280	139	Experimenting with a global decision tree for state clustering in automatic speech recognition systems	claim
281	139	Audio-assisted trajectory estimation in non-overlapping multi-camera networks	claim
282	139	Jointly optimized mode decisions in redundant video streaming	claim
283	139	On separating glottal source and vocal tract information in telephony speaker verification	claim
284	139	High-rate distributed multi-source cooperation using complex field coding	claim
285	139	Video coding using Variable Block-Size Spatially Varying Transforms	claim
286	139	On the recovery of nonnegative sparse vectors from sparse measurements inspired by expanders	claim
287	139	A new perceptual quality metric for compressed video	claim
288	138	Small-group learning projects to make signal processing more appealing: From speech processing to OFDMA synchronization	claim
289	138	Subspace tracking of fast time-varying channels in precoded MIMO-OFDM systems	claim
290	138	Instantaneous frequency rate estimation for high-order polynomial-phase signal	claim
291	138	Applying improved spectral modeling for High Quality voice conversion	claim
292	138	Game theory for precoding in a multi-user system: Bargaining for overall benefits	claim
293	138	View-invariant tensor null-space representation for multiple motion trajectory retrieval and classification	claim
294	137	On the use of Bayesian modeling for predicting noise reduction performance	claim
295	137	Voice conversion for various types of body transmitted speech	claim
296	137	Document reconstruction using dynamic programming	claim
297	137	Bounded conditional mean imputation with Gaussian mixture models: A reconstruction approach to partly occluded features	claim
298	137	Data hiding in hard-copy text documents robust to print, scan and photocopy operations	claim
299	137	Generic invertibility of multidimensional FIR multirate systems and filter banks	claim
300	137	Graphical Models: Statistical inference vs. determination	claim
301	137	GTD-based transceivers for decision feedback and bit loading	claim
302	137	Modelling the neurovascular habituation effect on fMRI time series	claim
303	136	Optimizing segment label boundaries for statistical speech synthesis	claim
304	136	Robust modeling of musical chord sequences using probabilistic N-grams	claim
305	136	Modelling the prepausal lengthening effect for speech recognition: a dynamic Bayesian network approach	claim
306	136	A multistage approach for blind separation of convolutive speech mixtures	claim
307	136	Real-time dynamic MR image reconstruction using Kalman Filtered Compressed Sensing	claim
308	136	Emotion recognition from speech: Putting ASR in the loop	claim
309	136	Robust discriminative keyword spotting for emotionally colored spontaneous speech using bidirectional LSTM networks	claim
310	136	Efficacy of a constantly adaptive language modeling technique for web-scale applications	claim
311	136	A variational EM algorithm for learning eigenvoice parameters in mixed signals	claim
312	135	Motion effect modeling in multipath configuration using warping based lag-Doppler filtering	claim
313	135	The effect of formant trajectories and phoneme durations on vowel intelligibility	claim
314	135	Fusing short term and long term features for improved speaker diarization	claim
315	135	Sparse source separation from orthogonal mixtures	claim
316	135	Combining discriminative re-ranking and co-training for parsing Mandarin speech transcripts	claim
317	135	A multiplicative weight perturbation scheme for distributed beamforming in wireless relay networks with 1-bit feedback	claim
318	135	Room impulse response shortening with infinity-norm optimization	claim
319	135	Multi-foveation filtering	claim
320	134	Unsupervised equalization of Lombard effect for speech recognition in noisy adverse environment	claim
321	134	Scalable superwideband extension for wideband coding	claim
322	134	A generalized family of parameter estimation techniques	claim
323	134	High-level feature extraction using SVM with walk-based graph kernel	claim
324	134	A MIMO channel perturbation analysis for robust bit loading	claim
325	134	Qualitative analysis of rotational modes within three dimensional empirical mode decomposition	claim
326	133	L1 regularized super-resolution from unregistered omnidirectional images	claim
327	133	A study on multilingual acoustic modeling for large vocabulary ASR	claim
328	133	Ranging energy optimization for robust sensor positioning	claim
329	133	A differential motion estimation method for image interpolation in distributed video coding	claim
330	133	Applications of complex augmented kernels to wind profile prediction	claim
331	132	Using fourier-based shape alignment to add geometric prior to snakes	claim
332	132	On robustness of coupled adaptive filters	claim
333	132	Query by tapping system based on alignment algorithm	claim
334	132	Extensions of absolute discounting (Kneser-Ney method)	claim
335	132	Gaussian Backend design for open-set language detection	claim
336	131	Convex analysis based minimum-volume enclosing simplex algorithm for hyperspectral unmixing	claim
337	131	Cross-layer optimization of wireless fading ad-hoc networks	claim
338	131	Periodic event detection and recognition in video	claim
339	131	Higher dimensional consensus algorithms in sensor networks	claim
340	131	The gigavision camera	claim
341	131	A wavelet-based quadratic extension method for image deconvolution in the presence of poisson noise	claim
342	131	Perceptual quality based packet dropping for generalized video GOP structures	claim
343	130	A nonparametric test for stationarity based on local Fourier analysis	claim
344	130	Spoken term detection using fast phonetic decoding	claim
345	130	An analytical approach to sound field reproduction with a movable sweet spot using circular distributions of loudspeakers	claim
346	130	Tampering identification using Empirical Frequency Response	claim
347	129	Training and adapting MLP features for Arabic speech recognition	claim
348	129	Long-time span acoustic activity analysis from far-field sensors in smart homes	claim
349	129	The effectiveness of histogram equalization on environmental model adaptation	claim
350	129	Blind system identification for speech dereverberation with Forced Spectral Diversity	claim
351	129	Layers and layer interfaces in wireless networks	claim
352	129	A Bayesian approach to HMM-based speech synthesis	claim
353	128	Multi-channel audio segmentation for continuous observation and archival of large spaces	claim
354	128	Sparse Decomposition over non-full-rank dictionaries	claim
355	128	Bayesian sparse image reconstruction for MRFM	claim
356	128	Estimating multiple transmitter locations from power measurements at multiple receivers	claim
357	128	Cross-lingual speech recognition under runtime resource constraints	claim
358	128	Target detection using incremental learning on single-trial evoked response	claim
359	128	On perfect channel identifiability of semiblind ML detection of orthogonal space-time block coded OFDM	claim
360	128	A performance-weighted mixture of LMS filters	claim
361	127	Low bitrate audio coding using generalized adaptive gain shape vector quantization across channels	claim
362	127	Using complex-valued ICA to efficiently combine radar polarimetric data for target detection	claim
363	127	Maximum-likelihood estimation of autoregressive models with conditional independence constraints	claim
364	127	Detecting real life anger	claim
365	127	Testing fractal connectivity in multivariate long memory processes	claim
366	127	Part-of-speech histograms for genre classification of text	claim
367	127	High improvement of speaker identification and verification by combining MFCC and phase information	claim
368	126	A scale transform based method for rhythmic similarity of music	claim
369	126	Sparse decomposition of mixed audio signals by basis pursuit with autoregressive models	claim
370	126	Exploiting user feedback for language model adaptation in meeting recognition	claim
371	126	Anti-collusion fingerprinting with scalar Costa scheme (SCS) and colluder weight recovery	claim
372	126	Posterior features applied to speech recognition tasks with user-defined vocabulary	claim
373	126	Frequency invariant MVDR beamforming without filters and implementation using MIMO radar	claim
374	126	Design of robust superdirective beamformers as a convex optimization problem	claim
375	126	Directed Markov Stationary Features for visual classification	claim
376	126	Enhanced error resilience of video communications for burst losses using an extended ROPE algorithm	claim
377	125	Wrapping snakes for improved lip segmentation	claim
378	125	Psychoacoustically constrained and distortion minimized speech enhancement algorithm	claim
379	125	Exploiting T-junctions for depth segregation in single images	claim
380	125	Incorporating monolingual corpora into bilingual latent semantic analysis for crosslingual LM adaptation	claim
381	125	Music emotion ranking	claim
382	125	Polyphonic musical instrument recognition based on a dynamic model of the spectral envelope	claim
383	125	Fast belief propagation process element for high-quality stereo estimation	claim
384	124	Restoring punctuation and capitalization in transcribed speech	claim
385	124	Optimal parameter estimation for model-based quantization	claim
386	124	Extended VTS for noise-robust speech recognition	claim
387	124	Performance bounds of MIMO receivers in the presence of radio frequency interference	claim
388	124	Transcoding based robust streaming of compressed video	claim
389	123	Combining mixture weight pruning and quantization for small-footprint speech recognition	claim
390	123	A low-complexity spectro-temporal based perceptual model	claim
391	123	Spatio-temporal adaptive detector in non-homogeneous and low-rank clutter	claim
392	123	Impact of novel sources on content-based image and video retrieval	claim
393	123	Unsupervised acoustic and language model training with small amounts of labelled data	claim
394	123	Improving acoustic speaker verification with visual Body-Language features	claim
395	123	Transceiver design with vector perturbation technique and iterative power loading	claim
396	123	A frequency hopping spread spectrum transmission scheme for uncoordinated cognitive radios	claim
397	123	Low-delay scheduling for Grassmannian beamforming with a SINR constraint	claim
398	122	Fast LCD motion deblurring by decimation and optimization	claim
399	122	Bandwidth-efficient cache-based motion compensation architecture with DRAM-friendly data access control	claim
400	122	Block Jacobi-type methods for non-orthogonal joint diagonalisation	claim
401	122	Consistent recovery of stimuli encoded with a neural ensemble	claim
402	122	Acoustic compensation methods for body transmitted speech conversion	claim
403	121	An EM algorithm for SCFG in formal syntax-based translation	claim
404	121	Underdetermined audio source separation from anechoic mixtures with long time delay	claim
405	121	Improving multi-lattice alignment based spoken keyword spotting	claim
406	121	A joint decoding algorithm for multiple-example-based addition of words to a pronunciation lexicon	claim
407	121	SkellamShrink: Poisson intensity estimation for vector-valued data	claim
408	121	Video event detection and summarization using audio, visual and text saliency	claim
409	121	Joint estimation of short-term and long-term predictors in speech coders	claim
410	121	Spoken dialog strategy based on understanding graph search	claim
411	120	Rate-constrained distributed distance testing and its applications	claim
412	120	Modeling of contours in wavelet domain for generalized lifting image compression	claim
413	120	On the phonetic information in ultrasonic microphone signals	claim
414	120	A sum-of-products model for effective coherent modulation filtering	claim
415	120	Discriminative training of hierarchical acoustic models for large vocabulary continuous speech recognition	claim
416	120	A new stochastic estimator for tremor frequency tracking	claim
417	120	Self-optimizing scheme for active noise and vibration control	claim
418	120	Neighborhood gossip: Concurrent averaging through local interference	claim
419	119	WEB-derived pronunciations	claim
420	119	Multichannel speech enhancement using convolutive transfer function approximation in reverberant environments	claim
421	118	Using collective information in semi-supervised learning for speech recognition	claim
422	118	Joint MIMO radar waveform and receiving filter optimization	claim
423	118	Design of a morphological moving object signature and application to human identification	claim
424	118	Complete characterization of perfectly secure stego-systems with mutually independent embedding operation	claim
425	118	Rate distribution between model and signal for multiple descriptions	claim
426	118	Automatic voice assignment tool for Instant Casting movie System	claim
427	118	Cache-based integer motion/disparity estimation for quad-HD H.264/AVC and HD multiview video coding	claim
428	117	Using phase linearity in frequency-domain ICA to tackle the permutation problem	claim
429	117	A game-theoretic framework for multi-user multimedia rate allocation	claim
430	117	Quickest change detection in multiple on-off processes	claim
431	117	Distributed peer-to-peer beamforming for multiuser relay networks	claim
432	117	Dictionary learning for the sparse modelling of atrial fibrillation in ECG signals	claim
433	117	An efficient first-order method for l1 compression of images	claim
434	116	Sparse boosting	claim
435	116	Modal expansion of HRTFs: Continuous representation in frequency-range-angle	claim
436	116	Sparse decomposition of two dimensional signals	claim
437	116	Improved clustered hierarchical tandem system with bottom-up processing	claim
438	116	Joint optimization of the redundancy of multiple-description coders for multicast	claim
439	115	Q-SIFT: Efficient feature descriptors for distributed camera calibration	claim
440	115	Shrinkage estimation of high dimensional covariance matrices	claim
441	115	Voice search of structured media data	claim
442	115	Progressive lossless compression of medical images	claim
443	115	Parsimonious dictionary learning	claim
444	115	Outage-based designs for multi-user transceivers	claim
445	115	Tree configuration games for distributed stream mining systems	claim
446	114	Trajectory training considering global variance for HMM-based speech synthesis	claim
447	114	Mobile media search	claim
448	114	Real-time PDA-based recursive Fourier transform Implementation for cochlear implant applications	claim
449	114	Generalization of specialized on-the-fly composition	claim
450	114	Visual grouping by neural oscillators	claim
451	114	Combining frontend-based memory with MFCC features for Bandwidth Extension of narrowband speech	claim
452	114	From rule-based to statistical grammars: Continuous improvement of large-scale spoken dialog systems	claim
453	114	Phoneme cluster based state mapping for text-independent voice conversion	claim
454	114	Ensembles of landmark multidimensional scalings	claim
455	113	Robust video fingerprinting based on visual attention regions	claim
456	113	Distributed sensing of signals linked by sparse filtering	claim
457	113	Resource usage prediction for groups of dynamic image-processing tasks using Markov modeling	claim
458	113	Minimax design of adjustable FIR filters using 2D polynomial methods	claim
459	112	Training-based Bayesian MIMO channel and channel norm estimation	claim
460	111	Learning to maximize signal-to-noise ratio for reverberant speech segregation	claim
461	110	Detecting bandlimited audio in broadcast television shows	claim
462	108	Using a pitch-synchronous residual codebook for hybrid HMM/frame selection speech synthesis	claim
463	108	Phase-based alignment of two signals having partially overlapped spectra	claim
464	108	Polyphase interpretation of empirical image interpolation	claim
465	107	Visual saliency with side information	claim
466	107	Signal processing challenges for future wireless communications	claim
467	107	Progressive lossy-to-lossless coding of arbitrarily-sampled image data using the modified scattered data coding method	claim
468	107	Revisiting graphemes with increasing amounts of data	claim
469	107	One-pass multi-layer rate-distortion optimization for quality scalable video coding	claim
470	107	Estimating correspondence between multiple cameras using joint invariants	claim
471	106	Multi-flow attack resistant watermarks for network flows	claim
472	105	On a tradeoff between dereverberation and noise reduction using the MVDR beamformer	claim
473	105	You are fired! Nonverbal role analysis in competitive meetings	claim
474	105	Quality control of automatic labelling using HMM-based synthesis	claim
475	105	Role of head pose estimation in speech acquisition from distant microphones	claim
476	104	Using filter banks to enhance images for fluid lens cameras based on color correlation	claim
477	104	Posterior-based confidence measures for spoken term detection	claim
478	104	Generalized mutual interdependence analysis	claim
479	104	LSH banding for large-scale retrieval with memory and recall constraints	claim
480	104	On ICA of improper and noncircular sources	claim
481	104	Dense error correction via l1-minimization	claim
482	104	Robust word boundary detection in spontaneous speech using acoustic and lexical cues	claim
483	104	Monaural voiced speech segregation based on elaborate harmonic grouping strategy	claim
484	102	Speech rhythm guided syllable nuclei detection	claim
485	102	Incorporating spectral subtraction and noise type for unvoiced speech segregation	claim
486	102	GridNews: A distributed automatic Greek broadcast transcription system	claim
487	101	Spatial super-resolution of a diffusion field by temporal oversampling in sensor networks	claim
488	101	Relative pitch estimation of multiple instruments	claim
489	100	Co-adaptation: Adaptive co-training for semi-supervised learning	claim
490	100	Multi-layer ratio Semi-Definite Classifiers	claim
491	99	Plane-wave decomposition of a sound scene using a cylindrical microphone array	claim
492	99	Risk-distortion analysis for video collusion attack	claim
493	97	On the role of localization cues in binaural segregation of reverberant speech	claim
494	95	An iterative approach to monaural musical mixture de-soloing	claim
495	94	On the error exponents for detecting randomly sampled noisy diffusion processes	claim
496	94	Narrowband interference mitigation in BICM OFDM systems	claim
497	91	A quality prediction model for JPEG2000-based color images	claim
498	85	Expectation-maximization algorithm for multi-pitch estimation and separation of overlapping harmonic spectra	claim
499	85	Facial age estimation by multilinear subspace analysis	claim
500	74	Single and multichannel sampling of bilevel polygons using exponential splines	claim

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