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Speech enhancement for hearing aids
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Speech Enhancement For Hearing Aids Using A Microphone Array
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Speech enhancement for hearing aids using adaptive beamformers
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Speech enhancement for hearing instruments: Enabling communication in adverse conditions
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Speech Enhancement for Listeners With Hearing Loss Based on a Model for Vowel Coding in the Auditory Midbrain
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Speech enhancement for mobile phones based on the imparity of two-microphone signals
7
Speech enhancement for mobile telephony
8
Speech enhancement for mobile telephony based on non-uniformly spaced frequency resolution
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Speech enhancement for nonstationary noise environment
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Speech enhancement for non-stationary noise environment by adaptive wavelet packet
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Speech Enhancement for Non-Stationary Noise Environments
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Speech enhancement for nonstationary noises by wavelet packet transform and adaptive noise estimation
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Speech enhancement for pathological voice using time-frequency trajectory excitation modeling
14
Speech enhancement for robust speech recognition in car environments using Griffiths-Jim ANC based on two-paired microphones
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Speech enhancement for telephony name speech recognition
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Speech enhancement for vehicle hands-free mobile telephony
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Speech Enhancement Iby Simulation Of Cocktail Party Effect With Neural Network Controlled Iterative Filter
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Speech enhancement in a Bayesian framework
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Speech enhancement in car noise envoronment based on an analysis-synthesis approach using harmonic noise model
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Speech Enhancement in Functional MRI Environment using Adaptive Sub-Band Algorithms
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Speech enhancement in harsh noisy environment using analytic decomposition of speech signal in critical bands
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Speech enhancement in hearing aids - from noise suppression to rendering of auditory scenes
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Speech enhancement in joint time-frequency domain based on real-valued discrete gabor transform
24
Speech enhancement in modulation domain using codebook-based speech and noise estimation
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Speech enhancement in noise and within face mask (microphone array approach)
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Speech enhancement in noisy environment using voice activity detection and wavelet thresholding
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Speech Enhancement in Noisy Environments for Video Retrieval
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Speech enhancement in presence of diffuse background noise: Why using blind signal extraction?
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Speech enhancement in the communication between vehicles
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Speech enhancement in transient noise environment using diffusion filtering
31
Speech enhancement in wireless digital communication via heuristic rules and image relaxation techniques
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Speech enhancement method based on spectral filtering utilizing asynchronous signals with embedded timecode over TCP/IP based network
33
Speech enhancement models suited for speech recognition using composite source and wavelet decomposition model
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Speech enhancement of color noise using empirical mode decomposition
35
Speech enhancement of non-stationary noise based on controlled forward moving average
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Speech enhancement of voice produced by an electronic larynx
37
Speech enhancement system based on lattice filter and system identification
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Speech enhancement through binaural negative filtering
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Speech enhancement through nonlinear adaptive source separation methods
40
Speech enhancement to overcome the effect of near-end noise in mobile phones using psychoacoustics
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Speech Enhancement Under a Combined Stochastic-Deterministic Model
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Speech Enhancement Under Aviation Noise
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Speech Enhancement Under Low SNR Conditions Via Noise Estimation Using Sparse and Low-Rank NMF with Kullback–Leibler Divergence
44
Speech enhancement using 2-D Fourier transform
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Speech enhancement using a constrained iterative sinusoidal model
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Speech enhancement using a frame adaptive gain function for Wiener filtering
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Speech enhancement using a frequency-specific composite Wiener function
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Speech enhancement using a joint MAP estimation of LP parameters
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Speech enhancement using a joint map estimator with Gaussian mixture model for (non-)stationary noise
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Speech enhancement using a Kalman-based normalized LMS algorithm
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Speech enhancement using a masking threshold constrained Kalman filter and its heuristic implementations
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Speech enhancement using a minimum mean-square error log-spectral amplitude estimator
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Speech enhancement using a minimum-mean square error short-time spectral amplitude estimator
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Speech enhancement using a mixture-maximum model
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Speech Enhancement Using a MMSE Short Time Spectral Amplitude Estimator with Laplacian Speech Modeling
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Speech enhancement using a modified Kalman filter based on complex linear prediction and supergaussian priors
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Speech enhancement using a noncausal a priori SNR estimator
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Speech enhancement using a nonlinear neural switched Griffiths-Jim beamformer
59
Speech enhancement using a perceptually constrained TQSVD-based approach incorporating PSO techniques
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Speech enhancement using a pitch predictive model
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Speech enhancement using a remote wireless microphone
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Speech enhancement using a robust Kalman filter post-processor in the modulation domain
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Speech enhancement using a soft-decision noise suppression filter
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Speech enhancement using a state-based transform model
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Speech Enhancement Using a Switching Kalman Filter with a Perceptual Post-Filter
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Speech Enhancement Using a Technique of Adaptive Bias Suppression
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Speech enhancement using a ternary-decision based filter
68
Speech enhancement using a variable suppression rule in Hilbert domain
69
Speech enhancement using a wiener filtering under signal presence uncertainty
70
Speech Enhancement using Adaptive Empirical Mode Decomposition
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Speech enhancement using adaptive MMSE estimator under signal presence uncertainty and Laplacian prior
72
Speech enhancement using adaptive neuro-fuzzy filtering
73
Speech enhancement using adaptive wavelet shrinkage
74
Speech enhancement using Affine Projection Algorithm in subband
75
Speech enhancement using an adaptive gain equalizer with frequency dependent parameter settings
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Speech enhancement using an Improved MMSE estimator with Laplacian prior
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Speech enhancement using a-priori information with classified noise codebooks
78
Speech enhancement using array signal processing with 4th-order cumulant
79
Speech enhancement using auditory spectral attenuation
80
Speech enhancement using auditory-based transform
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Speech enhancement using backward adaptive filtering algorithm: Variable step-sizes approaches
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Speech enhancement using beamforming and non negative matrix factorization for robust speech recognition in the CHiME-3 challenge
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Speech enhancement using bionic wavelet transform and adaptive threshold function
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Speech enhancement using blind source separation and two-channel energy based speaker detection
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Speech enhancement using combinational adaptive filtering
86
Speech enhancement using convex combination of fractional least-mean-squares algorithm
87
Speech enhancement using convolutive non-negative matrix factorization with multiple dictionaries
88
Speech enhancement using cross-correlation compensated multi-band Wiener filter combined with constrained perceptual weighting filter
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Speech enhancement using CSS-based array processing
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Speech enhancement using Discriminative Random Fields
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Speech enhancement using excitation source information
92
Speech enhancement using Fast Hartley Transform
93
Speech Enhancement Using Gain Function of Noisy Power Estimates and Linear Regression
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Speech Enhancement Using Gaussian Scale Mixture Models
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Speech enhancement using generalized least absolute deviation estimation
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Speech enhancement using generalized maximum a posteriori spectral amplitude estimator
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Speech Enhancement Using Generative Dictionary Learning
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Speech Enhancement Using Harmonic Emphasis and Adaptive Comb Filtering
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Speech Enhancement Using Harmonic Regeneration
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Speech enhancement using harmonic regeneration
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Speech enhancement using harmonic regeneration with improved wavelet based a-priori signal to noise ratio estimator
102
Speech enhancement using hybrid Generalized Sidelobe Canceller and spectral estimator
103
Speech enhancement using ICA with Bessel features
104
Speech Enhancement Using Inter-Frame Dependence in DCT Domain
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Speech enhancement using intra-frame dependency in DCT domain
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Speech enhancement using Kalman Filter for white, random and color noise
107
Speech enhancement using Kalman filters for restoration of short-time DFT trajectories
108
Speech enhancement using kernel adaptive filtering method
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Speech Enhancement Using Masking of Noisy Power Estimates
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Speech enhancement using microphone array in moving vehicle environment
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Speech enhancement using microphone array neural switched Griffiths-Jim beamformer
112
Speech enhancement using minimum mean square error filter and spectral subtraction filter
113
Speech enhancement using minimum mean-square error estimation and a post-filter derived from vector quantization of clean speech
114
Speech Enhancement Using MMSE Estimation and Spectral Subtraction Methods
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Speech enhancement using MMSE short time spectral estimation with gamma distributed speech priors
116
Speech enhancement using modified IMCRA and OMLSA methods
117
Speech enhancement using modified magnitude and phase spectra
118
Speech enhancement using modified minimun mean square error short-time spectral amplitude estimator [minimun read minimum]
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Speech enhancement using multiple soft constrained subband beamformers and non-coherent technique
120
Speech enhancement using multi-pulse excited linear prediction system
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Speech enhancement using noise estimation based on dynamic quantile tracking for hearing impaired listeners
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Speech enhancement using non negative matrix factorization and enhanced NMF
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Speech enhancement using nonlinear microphone array with complementary beamforming
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Speech enhancement using nonlinear microphone array with noise adaptive complementary beamforming
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Speech enhancement using non-linear prediction
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Speech enhancement using normalized cumulant-based adaptive algorithm for transient noise
127
Speech enhancement using nullspace-based sound field control for barge-in free spoken dialogue interface
128
Speech enhancement using optimal non-linear spectral amplitude estimation
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Speech enhancement using orthogonal matching pursuit algorithm
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Speech Enhancement Using Particle Swarm Optimization Techniques
131
Speech enhancement using PCA and variance of the reconstruction error in distributed speech recognition
132
Speech enhancement using perceptual multi_band Wiener filter
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Speech Enhancement Using Perceptual Wavelet Thresholding with the Ephraim and Malah Noise Suppressor and Auditory Masking
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Speech enhancement using perceptual Wiener filter combined with unvoiced speech — A new Scheme
135
Speech Enhancement Using Perceptual-Decision-Directed Approach
136
Speech enhancement using perceptually-constrained gain factors in critical-band-wavelet-packet transform
137
Speech enhancement using pre-image iterations
138
Speech enhancement using pre-processing
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Speech enhancement using psychoacoustic criteria
140
Speech enhancement using resonator filterbanks
141
Speech enhancement using robust weighting factors for critical-band-wavelet-packet transform
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Speech enhancement using segmental nonnegative matrix factorization
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Speech enhancement using sexual reproduction-based PSO
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Speech enhancement using soft thresholding with DCT-EMD based hybrid algorithm
145
Speech Enhancement using Source Features and Group Delay Analysis
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Speech Enhancement Using Spatial Processing and Modified Excitation Source for Underwater Speech Communication
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Speech enhancement using speaker dependent codebooks
148
Speech enhancement using spectral envelope side information
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Speech enhancement using spectral over-subtraction and residual noise reduction
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Speech enhancement using spectral subtraction and cascaded-median based noise estimation for hearing impaired listeners
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Speech Enhancement Using Spectral Subtraction Based on a Modified Noise Minimum Statistics Estimation
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Speech enhancement using spectral weighting with EMD-based post filtering
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Speech enhancement using square microphone array for mobile devices
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Speech enhancement using state dependent dynamical system model
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Speech enhancement using sub-band wiener filter with pitch synchronous analysis
156
Speech enhancement using temporal masking in presence of near-end noise
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Speech enhancement using the bispectrum
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Speech enhancement using the combination of adaptive wavelet threshold and spectral subtraction based on wavelet packet decomposition
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Speech enhancement using the constrained-optimization technique
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Speech enhancement using the dual excitation speech model
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Speech enhancement using the microphone array and short-term spectral amplitude estimator
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Speech enhancement using the multistage Wiener filter
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Speech enhancement using the sparse code shrinkage technique
164
Speech enhancement using transfer function ratio beamformer and matched filter array
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Speech enhancement using transient components in frequency domain
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Speech Enhancement Using Transient Speech Components
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Speech enhancement using vector quantization and a formant distance measure
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Speech enhancement using voice source models
169
Speech enhancement using warped discrete cosine transform
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Speech enhancement usinga modulation domain Kalman filter post-processor with a Gaussian Mixture noise model
171
Speech enhancement via adaptive eigenfiltering
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Speech enhancement via covariance estimation using Hermitian angle in adaptive beamforming
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Speech enhancement via frequency bandwidth extension using line spectral frequencies
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Speech enhancement via low-rank matrix decomposition and image based masking
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Speech enhancement via separation of sources from co-located microphone recordings
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Speech enhancement via sparse coding with ideal binary mask
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Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation
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Speech enhancement with a low-complexity online source number estimator using distributed arrays
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Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller
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Speech enhancement with ad-hoc microphone array using single source activity
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Speech Enhancement with EMD and Hurst-Based Mode Selection
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Speech enhancement with harmonic synthesis
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Speech Enhancement With Inventory Style Speech Resynthesis
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Speech enhancement with Kalman filtering the short-time DFT trajectories of noise and speech
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Speech enhancement with MAP estimation and ICA-based speech features
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Speech enhancement with masking properties in eigen-domain for colored noise
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Speech enhancement with missing data techniques using recurrent neural networks
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Speech enhancement with noise parameter estimated by a sequential Monte Carlo method
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Speech enhancement with reduction of noise components in the wavelet domain
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Speech enhancement with sparse coding in learned dictionaries
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Speech enhancement with zero replacement signal at low SNR
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Speech Enhancement, Gain, and Noise Spectrum Adaptation Using Approximate Bayesian Estimation
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Speech ermotion recognition separately from voiced and unvoiced sound for emotional interaction robot
194
Speech estimation from body-conducted speech with differential acceleration
195
Speech event detection by non negative matrix deconvolution
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Speech event detection using SVM and NMD
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Speech excitation modelling for low bit speech coding
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Speech expressiveness: Modeling and implementing the expressive impact of typographic and punctuation marks for textual inputs
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Speech extraction based on ICA and audio-visual coherence
200
Speech extraction method based on multiple reference signals ICA algorithm
201
Speech feature analysis and spectrum conversion from children to young adults
202
Speech Feature Analysis Using Temporal Linear Embedding
203
Speech feature analysis using variational Bayesian PCA
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Speech feature compensation in multiple model based speech recognition system using vts-based environmental parameter estimation
205
Speech feature denoising and dereverberation via deep autoencoders for noisy reverberant speech recognition
206
Speech Feature Estimation Under the Presence of Noise with a Switching Linear Dynamic Model
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Speech feature extracted from adaptive wavelet for speech recognition
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Speech feature extraction method of improved KPCA
209
Speech feature extraction method representing periodicity and aperiodicity in sub bands for robust speech recognition
210
Speech feature extraction of cochlear implants on the basis of auditory perception wavelet transform
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Speech feature extraction of KPCA based on kernel fuzzy K-means Clustering
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Speech feature extraction using independent component analysis
213
Speech Feature Mapping Based on Switching Linear Dynamic System
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Speech Feature Smoothing for Robust ASR
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Speech features found in a continuous high order statistical analysis of speech
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Speech for technical communicators: A bibliography
217
Speech formant frequency and bandwidth tracking using multiband energy demodulation
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Speech formant frequency and pitch estimation using instantaneous complex frequency
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Speech formant trajectory pattern recognition using multiple-order pole-focused LPC analysis
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Speech Formation Objectivisation Methods Development and Analysis for Correct Phonation Diagnostics
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Speech from stored data
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Speech generation for Albanian written texts
223
Speech generation through waveform synthesis
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Speech identification using a sequence-hased heuristic
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Speech in digital communication systems
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Speech in human system interaction
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Speech in Noisy Environments: robust automatic segmentation, feature extraction, and hypothesis combination
228
Speech in the VQE device environment
229
Speech Independent Component Analysis
230
Speech Information Hiding Method Based on Itakura-Saito Measure and Psychoacoustic Model
231
Speech Information Hiding Using Fractional Cosine Transform
232
Speech Information Processing: Theory and Applications [Scanning the Issue]
233
Speech Input Equipment
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Speech input output interfaces as a disruptive influence in transport control systems: a case against direct voice input/output interfaces in command and control
235
Speech input systems for broadcasting transmitters
236
Speech Input/Output
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Speech input-output system in Indian farming sector
238
Speech inputs to safety logic (SISL)
239
Speech inputs to surface safety logic systems
240
Speech intelligibility and MTF in non-exponential decay fields
241
Speech intelligibility enhancement in a power generating noise environment
242
Speech Intelligibility Enhancement using Tunable Equalization Filter
243
Speech Intelligibility Evaluation of Low Bit Rate Speech Codecs
244
Speech intelligibility improvement of cochlear implant using release of masking
245
Speech Intelligibility Improvement Using the Constraints on Speech Distortion and Noise Over-estimation
246
Speech intelligibility in the presence of cross-channel spectral asynchrony
247
Speech intelligibility measures for speech enhancement under nonstationary acoustic noise
248
Speech intelligibility of ideal binary masked mixtures
249
Speech intelligibility of VoIP to PSTN interworking - a key index for the QoS
250
Speech Intelligibility Prediction Based on Mutual Information
251
Speech intelligibility with a cochlear implant sound-processing scheme based on loudness models
252
Speech interaction in virtual reality
253
Speech Interaction with the Internet -- A User Study
254
Speech interactive agent system for car navigation using embedded ASR/TTS and DSR
255
Speech interface ASIC of SOC architecture for embedded application
256
Speech interface VLSI for car applications
257
Speech interface: an enhancer to the acceptance of m-commerce applications
258
Speech interfaces for mobile communications
259
Speech interfaces in real-time control systems
260
Speech Interpolation in Digital Transmission Systems
261
Speech interruption effects in multi-modal tasks
262
Speech into Sign Language Statistical Translation System for Deaf People
263
Speech inversion: Benefits of tract variables over pellet trajectories
264
Speech kurtosis estimation from observed noisy signal based on generalized Gaussian distribution prior and additivity of cumulants
265
Speech labeling and the most frequent phrase extraction using same section in a presentation speech
266
Speech Length Threshold in Forensic Speaker Comparison by Using Long-Term Cumulative Formant (LTCF) Analysis
267
Speech LSF quantization with rate independent complexity, bit scalability and learning
268
Speech Magnitude-Spectrum Information-Entropy (MSIE) for Automatic Speech Recognition in Noisy Environments
269
Speech Maker: text-to-speech synthesis based on a multi-level, synchronized data structure
270
Speech measurements using a laser Doppler vibrometer sensor: Application to speech enhancement
271
Speech modeling and voiced/unvoiced/mixed/silence speech segmentation with fractionally Gaussian noise based models
272
Speech modeling based on committee-based active learning
273
Speech modeling via model reduction
274
Speech modeling with time varying amplitude and phase
275
Speech Modelingwith Magnitude-Normalized Complex Spectra and Its Application to Multisensory Speech Enhancement
276
Speech Modelling Based On Generalized Gaussian Probability Density Functions
277
Speech Modelling Based on Phone Statistics
278
Speech modelling by model-order reduction: SNR behaviour
279
Speech modelling by non-stationary partials with time varying amplitude and frequency
280
Speech modelling using cepstral-time feature matrices and hidden Markov models
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Speech modelling using cepstral-time feature matrices in hidden Markov models
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Speech modelling using phoneme segmentation and modified weighted levenshtein distance
283
Speech modification by selective Fourier-Bessel series expansion of speech signals
284
Speech Modification for More Regularity and Pitch Estimation Improvement
285
Speech modification for prosody conversion in expressive Marathi text-to-speech synthesis
286
Speech modifications algorithms used for training language learning-impaired children
287
Speech monitoring of infective laryngitis
288
Speech morphing by gradually changing spectrum parameter and fundamental frequency
289
Speech multiplexing in wireless PCS employing resource auction multiple access (RAMA)
290
Speech music discrimination using class-specific features
291
Speech Noise Estimation using Enhanced Minima Controlled Recursive Averaging
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Speech noise reduction by means of multi-signal minimum-cross-entropy spectral analysis
293
Speech Noise Reduction System Using MDFT Pair - Formulation of Noise Subtraction Coefficient -
294
Speech noise reduction using sequential spectrum detection based on modified DFT pair
295
Speech nonlinearities, modulations, and energy operators
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Speech notes: How and when to use them
297
Speech of Acceptance for 1948 Fellows of the IRE
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Speech of Mr. Clifford C. Paterson at the Government banquet, 25th September, 1931
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Speech output from a computer-controlled water-supply network
300
Speech output in HCI technologies
301
Speech output systems and the Captain Kirk problem
302
Speech over VoIP Networks: Advanced Signal Processing and System Implementation
303
Speech overlap detection and attribution using convolutive non-negative sparse coding
304
Speech overlap detection using convolutive non-negative sparse coding: New improvements and insights
305
Speech Overlap Detection using Spectral Features and its Application in Speech Indexing
306
Speech Packet Concealment Techniques Based on Time-Scale Modification for VoIP
307
Speech parameter estimation by time-weighted-error Kalman filtering
308
Speech parameter estimation using a vocal tract/Cord model
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Speech parameter extraction in noisy environment using a masking model
310
Speech parameter generation algorithms for HMM-based speech synthesis
311
Speech parameter generation considering LSP ordering property for HMM-based speech synthesis
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Speech parameter generation from HMM using dynamic features
313
Speech parameter selection for emotional stress characterization in women
314
Speech parameters vector based on arithmetic fourier transform
315
Speech pathological study of epinosic patient using artificial neural network
316
Speech pattern classification using Large Geometric Margin Minimum Classification Error training
317
Speech pattern generation and recognition using a personal computer
318
Speech pause detection for noise spectrum estimation by tracking power envelope dynamics
319
Speech perception and cochlear signal processing [Life Sciences]
320
Speech perception based algorithm for the separation of overlapping speech signal
321
Speech perception via the tactile mode: Progress report
322
Speech perception, production, and the sensorimotor mu rhythm
323
Speech perception: Single trial analysis of the N1/P2 complex of unimodal and audiovisual evoked responses
324
Speech perceptual abilities of children with specific reading difficulty (dyslexia)
325
Speech performance of adaptive transceivers for PCN
326
Speech pitch detection in noisy environment using multi-rate adaptive lossless FIR filters
327
Speech pitch detection using short-time energy
328
Speech Pitch Frequency as an Emotional State Indicator
329
Speech Pitch Period Detection Algorithm Based on Wavelet Transform and Spacial Correlation Function
330
Speech pitch period estimation using circular AMDF
331
Speech Pitch Shifting using Complex Continuous Wavelet Transform
332
Speech planning of an anthropomorphic talking robot for consonant sounds production
333
Speech playback geometry for smart homes
334
Speech playout buffering based on a simplified version of the ITU-T E-model
335
Speech Plot Display for the Deaf-mute based on Combined Characters Encoding of Speech Signal
336
Speech power estimation with a truncated normal distribution
337
Speech prediction using higher order neural networks
338
Speech prediction using recurrent neural networks
339
Speech Predictor Design for Diversity Communication Systems
340
Speech presence detection in the time-frequency domain using minimum statistics
341
Speech presence probability estimation based on integrated time-frequency minimum tracking for speech enhancement in adverse environments
342
Speech presence probability estimation based on temporal cepstrum smoothing
343
Speech privacy for modern mobile communication systems
344
Speech privacy technophobes need not apply
345
Speech probability distribution
346
Speech Processing
347
Speech processing aids for the deaf:An overview
348
Speech processing and retrieval in a personal memory aid system for the elderly
349
Speech processing and synthesis for assessing vocal disorders
350
Speech processing at BBN
351
Speech processing by a model of the auditory periphery
352
Speech processing by hierarchical segment classification
353
Speech processing by splicing of autocorrelation function
354
Speech processing committee
355
Speech processing for artificial tactile speech displays
356
Speech Processing for Cochlear Implants with the Discrete Wavelet Transform: Feasibility Study and Performance Evaluation
357
Speech processing for hearing aids for moderate bilateral sensorineural hearing loss
358
Speech processing for makhraj recognition
359
Speech processing for makhraj recognition: The design of adaptive filter for noise canceller
360
Speech processing in a high ambient noise environment
361
Speech processing in FPGA with C-to-RTL compiler technology
362
Speech processing in Java-based PC speech commanding application
363
Speech processing in mobile radio communications
364
Speech processing in packet communication systems
365
Speech Processing in Support of Human-Human Communication (Invited Paper)
366
Speech processing in the Pan-European digital cellular mobile telephone system
367
Speech processing in the pan-European digital mobile radio system (GSM)-system overview
368
Speech processing in the watermarked domain: Application in adaptive Acoustic Echo Cancellation
369
Speech processing model based on the cochlea temporal characteristics and neural networks with Gabor weights
370
Speech Processing on a Reconfigurable Analog Platform
371
Speech processing on a reconfigurable analog platform
372
Speech processing system by use of short-time crosscorrelation function
373
Speech processing systems
374
Speech processing technical committee meeting
375
Speech processing techniques and applications
376
Speech processing techniques for HF radio security
377
Speech processing technologies and telecommunications applications at NTT
378
Speech processing technology of Uyghur language
379
Speech Processing Using Higher Order Statistics
380
Speech processing using higher order statistics
381
Speech Processing Using Joint Features Derived from the Modified Group Delay Function
382
Speech processing using programmable VLSI
383
Speech processing utilizing the Kalman filter
384
Speech processing with a Boltzmann machine
385
Speech processing with a cortical representation of audio
386
Speech Processing with the Usage of Cosine Transform on the Basis of Synthetic DPCM
387
Speech processing with Walsh-Hadamard transforms
388
Speech processing: past and future
389
Speech processors for cochlear prostheses
390
Speech production and perception models and their applications to synthesis, recognition, and coding
391
Speech production by a mechanical model: Construction of a vocal tract and its control by neural network
392
Speech production modelling with variable glottal reflection coefficient
393
Speech production of an advanced talking robot based on human acoustic theory
394
Speech production-physics, models and prospective applications
395
Speech prosody control using weighted neural network ensembles
396
Speech quality and recognition rate improvement in car noise environments
397
Speech quality assessment
398
Speech quality assessment of compounded digital telecommunication systems; perceptual dimensions
399
Speech quality assessment using digital watermarking
400
Speech quality assessment using expert pattern recognition
401
Speech Quality Aware Admission Control for Fixed IEEE 802.16 Wireless MAN
402
Speech quality based on Arabic pronunciation using MFCC and LDA: Investigating the emphatic consonants
403
Speech quality based performance evaluation of a partially loaded GSM system
404
Speech Quality Enhancement Based on Sinusoidal Model Using Chebyshev Filter
405
Speech quality estimation in MIMO-OFDM systems
406
Speech Quality Estimation: Models and Trends
407
Speech Quality Evaluation and Benchmarking in Cellular Mobile Networks
408
Speech quality evaluation for mobile networks
409
Speech quality evaluation of hands-free telephones during double talk: New evaluation methodologies
410
Speech Quality Evaluation: A New Application of Digital Watermarking
411
Speech Quality Evaluation: A New Application of Digital Watermarking
412
Speech Quality Improvement Based on List Viterbi and Joint Source-Channel Decoding in UMTS
413
Speech quality improvement in ATM networks using an alternating-cell discarding strategy
414
Speech quality measure based on auditory scene analysis
415
Speech quality objective assessment using neural network
416
Speech Quality of IEEE 802.11b and 802.11g Wireless LANs
417
Speech Quality Prediction in VoIP Concatenating Multiple Markov-Based Channels
418
Speech quality prediction in VoIP using the extended E-model
419
Speech quality testing method of Cockpit Voice Recorder
420
Speech Quality While Roaming in Next Generation Networks
421
Speech rate change detection in martingale framework
422
Speech Rate Control for The Compound HearingAid Processing
423
Speech rate conversion without extraction of accurate pitch period
424
Speech rate determination by vowel detection on the modulated energy envelope
425
Speech Rate Estimation via Temporal Correlation and Selected Sub-Band Correlation
426
Speech Rating System through Space Mapping
427
Speech recognition
428
Speech recognition
429
Speech recognition
430
Speech recognition
431
Speech recognition - based control system for Drone
432
Speech recognition algorithm based on switched capacitor network system
433
Speech recognition and enhancement by a nonstationary AR HMM with gain adaptation under unknown noise
434
Speech recognition and frequency selectivity for hearing impaired listeners
435
Speech recognition and its application in voice-based robot control system
436
Speech recognition and noise
437
Speech Recognition and Spoken Language Understanding for Mobile Personal Assistants: A Case Study of "Shabette Concier"
438
Speech recognition and utterance verification based on a generalized confidence score
439
Speech Recognition Applied in Games
440
Speech recognition approach: Desktop items activation with comparative analysis
441
Speech Recognition as Feature Extraction for Speaker Recognition
442
Speech recognition based computer keyboard replacement for the Quadriplegics, Paraplegics, paralytics and amputees
443
Speech recognition based on a model of human auditory system
444
Speech recognition based on acoustically derived segment units
445
Speech recognition based on auditory wavelet packet filter
446
Speech recognition based on cooperative particle swarm optimizer wavelet neural network
447
Speech recognition based on fundamental functional principles of the brain
448
Speech recognition based on fuzzy vector quantization and fuzzy logic
449
Speech recognition based on genetic algorithm for training HMM
450
Speech recognition based on k-means clustering and neural network ensembles
451
Speech recognition based on Kohonen self-organizing feature maps and hybrid connectionist systems
452
Speech recognition based on lateral inhibition network auditory model
453
Speech recognition based on space diversity using distributed multi-microphone
454
Speech Recognition Based on Support Vector Machine and Error Correcting Output Codes
455
Speech recognition based on variable information rate model
456
Speech recognition based on wavelet packet transform and K-L expansion
457
Speech recognition based on zero crossing rate and energy
458
Speech recognition based wireless automation of home loads with fault identification for physically challenged
459
Speech Recognition by Classifying Speech Signals Based on the Fire Fly and Fuzzy
460
Speech recognition by combining pairwise discriminant time-delay neural networks and predictive LR-parser
461
Speech Recognition By Computer
462
Speech recognition by dynamic recurrent neural networks
463
Speech recognition by extended loop neural network
464
Speech recognition by indexing and sequencing
465
Speech recognition by machine: A review
466
Speech recognition by machine¿art or science?
467
Speech recognition by using the extended associative memory neural network (EAMNN)
468
Speech recognition chip for monosyllables
469
Speech recognition complexity reduction using decimation of cepstral time trajectories
470
Speech recognition engine adaptions for smart home dialogues
471
Speech Recognition Engineering Issues in Speech to Speech Translation System Design for Low Resource Languages and Domains
472
Speech recognition enhancement by psychoacoustic modeled noise suppression
473
Speech Recognition Enhancement Using Beamforming and a Genetic Algorithm
474
Speech Recognition error correction by using combinational measures
475
Speech recognition experiments using multi-span statistical language models
476
Speech recognition experiments with guided propagation
477
Speech recognition experiments with linear predication, bandpass filtering, and dynamic programming
478
Speech recognition features for EEG signal description in detection of neonatal seizures
479
Speech recognition features: Comparison studies on robustness against environmental distortions
480
Speech recognition for a distant moving speaker based on HMM composition and separation
481
Speech Recognition for a Humanoid with Motor Noise Utilizing Missing Feature Theory
482
Speech recognition for an information kiosk
483
Speech recognition for DARPA Communicator
484
Speech Recognition for Disabilities People
485
Speech recognition for game control
486
Speech Recognition for Illiterate Access to Information and Technology
487
Speech recognition for image animation and coding
488
Speech recognition for Japanese spoken language
489
Speech recognition for mixed speech and music by NMF using various cost functions and noise adaptive training methods
490
Speech recognition for resource deficient languages using frugal speech corpus
491
Speech recognition for spontaneously spoken German dialogues
492
Speech recognition for the disabled
493
Speech recognition for the Siemens EWSD public exchange
494
Speech Recognition for Turkish Broadcast News
495
Speech Recognition for Voice-Based Machine Translation
496
Speech recognition for wireless applications
497
Speech recognition from adaptive windowing PSD estimation
498
Speech recognition HMM training on reconfigurable parallel processor
499
Speech recognition in a dialog system for patient health monitoring
500
Speech recognition in a home environment using parallel decoding with GMM-based noise modeling
501
Speech recognition in a multiprocessor environment
502
Speech recognition in a neural network framework: discriminative training of Gaussian models and mixture densities as radial basis functions
503
Speech recognition in a noisy environment using a noise reduction neural network and a codebook mapping technique
504
Speech recognition in a reverberant environment using matched filter array (MFA) processing and linguistic-tree maximum likelihood linear regression (LT-MLLR) adaptation
505
Speech recognition in a smart home: Some experiments for telemonitoring
506
Speech recognition in a unification grammar framework
507
Speech recognition in adverse environments using lip information
508
Speech recognition in analog multichannel cochlear prostheses: initial experiments in controlling classifications
509
Speech Recognition in Aphasiac Rehabilitation Based on Wavelet
510
Speech Recognition in Human Mediated Translation Scenarios
511
Speech recognition in impulsive noise
512
Speech recognition in multiple languages and domains: the 2003 BBN/LIMSI EARS system
513
Speech Recognition in Multisource Reverberant Environments with Binaural Inputs
514
Speech recognition in noise using a projection-based likelihood measure for mixture density HMM´s
515
Speech recognition in noise using a self-structuring noise reduction model and hidden control models
516
Speech recognition in noisy car environment based on OSALPC representation and robust similarity measuring techniques
517
Speech recognition in noisy environment, issues and challenges: A review
518
Speech Recognition in Noisy Environments with Convolutional Neural Networks
519
Speech recognition in non-stationary adverse environments
520
Speech Recognition in Reverberant Environments Using Remote Microphones
521
Speech recognition in scale space
522
Speech recognition in the blind condition based on multiple directivity patterns using a microphone array
523
Speech recognition in the context of two-way immediate person-machine interaction
524
Speech recognition in the F-16 cockpit using principal spectral components
525
Speech recognition in the Informedia Digital Video Library: uses and limitations
526
Speech recognition in the question-answering system operated by conversational speech
527
Speech recognition in VODIS II
528
Speech recognition issues for directory assistance applications
529
Speech Recognition Method Based on Linear Descending Inertia Weight PSO Algorithm Optimizing SVM Kernel Parameters
530
Speech recognition method based on weighed autoregressive HMM
531
Speech recognition method using quantised LSP parameters in CELP-type coders
532
Speech recognition methods applied to biomedical signals processing
533
Speech recognition methods for speech therapy
534
Speech recognition modeling advances for mobile voice search
535
Speech Recognition Moves from Software to Hardware
536
Speech recognition network with SC neuron-type components
537
Speech recognition of a named entity
538
Speech recognition of a natural text read as isolated words
539
Speech recognition of aged voice in the AAL context: Detection of distress sentences
540
Speech recognition of broadcast news for the European Portuguese language
541
Speech recognition of deaf and hard of hearing people using hybrid neural network
542
Speech Recognition of Isolated Malayalam Words Using Wavelet Features and Artificial Neural Network
543
Speech recognition of Malayalam numbers
544
Speech recognition of monosyllables using hidden Markov model in VHDL
545
Speech recognition of multiple accented English data using acoustic model interpolation
546
Speech recognition of spontaneous, noisy speech using auxiliary information in Bayesian networks
547
Speech recognition on a distributed array processor
548
Speech Recognition on Code-Switching Among the Chinese Dialects
549
Speech recognition on DSP: issues on computational efficiency and performance analysis
550
Speech recognition on Mandarin Call Home: a large-vocabulary, conversational, and telephone speech corpus
551
Speech recognition on MPEG/Audio encoded files
552
Speech recognition on the DADO/DSP multiprocessor
553
Speech recognition open source tools for the semantic identification of the sentence
554
Speech recognition over Bluetooth ACL and SCO links: a comparison
555
Speech recognition over the Internet using Java
556
Speech recognition performance as an effective perceived quality predictor
557
Speech recognition performance assessments and available databases
558
Speech recognition performance on a voicemail transcription task
559
Speech recognition processors using fuzzy pattern matching
560
Speech Recognition Research on Uyghur Accent Spoken Language
561
Speech Recognition Standard Procedures, Error Recognition and Repair Strategies
562
Speech recognition strategy for stress-induced speech
563
Speech recognition system based improved DTW algorithm
564
Speech recognition system based on DSP and SVM
565
Speech Recognition System Based on Integrating Feature and HMM
566
Speech Recognition System Based on Visual Feature for the Hearing Impaired
567
Speech Recognition System Combination for Machine Translation
568
Speech recognition system for automatic telephone operator based on CSS architecture
569
Speech recognition system for embedded real-time applications
570
Speech Recognition System Using a Wavelet Packet and Synergetic Neural Network
571
Speech recognition system using adapted local trigonometric transforms
572
Speech Recognition Systems and Its Automatic Synthesis in Hardware
573
Speech recognition techniques
574
Speech recognition techniques using RBF networks
575
Speech Recognition Technology
576
Speech recognition technology in the ubiquitous/wearable computing environment
577
Speech recognition through phoneme segmentation and neural classification
578
Speech Recognition Tools for Mobile Phone - A Comparative Study
579
Speech recognition under additive noise
580
Speech recognition under noisy environments using multiple microphones based on asynchronous and intermittent measurements
581
Speech recognition using a cochlear model
582
Speech recognition using a sequential neural network
583
Speech recognition using a strong correlation assumption for the instantaneous spectra
584
Speech recognition using advanced HMM2 features
585
Speech recognition using an auditory model with pitch-synchronous analysis
586
Speech recognition using an enhanced FVQ based on a codeword dependent distribution normalization and codeword weighting by fuzzy objective function
587
Speech Recognition using Artificial Neural Networks
588
Speech recognition using artificial neural networks
589
Speech Recognition Using Augmented Conditional Random Fields
590
Speech recognition using autocorrelation analysis
591
Speech recognition using automatically derived acoustic baseforms
592
Speech Recognition using Back-Propagation Neural Networks
593
Speech recognition using blind source separation and dereverberation method for mixed sound of speech and music
594
Speech recognition using combined forward and backward Viterbi search
595
Speech recognition using dual-stage hierarchical structure: Gross and fine phonetic features
596
Speech recognition using dynamic features of acoustic subword spectra
597
Speech recognition using dynamic neural networks
598
Speech recognition using dynamic recognition neural network
599
Speech Recognition using Dynamic Time Warping
600
Speech recognition using dynamic time warping with neural network trained templates
601
Speech recognition using dynamic transformation of phoneme templates depending on acoustic/phonetic environments
602
Speech recognition using dynamical model of speech production
603
Speech Recognition using FHMMS Robust Against Nonstationary Noise
604
Speech recognition using filter-bank features
605
Speech recognition using frequency transformations
606
Speech recognition using fuzzy logic
607
Speech recognition using hidden Markov model decomposition and a general background speech model
608
Speech Recognition Using Hidden Markov Model with MFCC-Subband Technique
609
Speech recognition using hidden Markov models and artificial neural networks
610
Speech recognition using hidden Markov models with polynomial regression functions as nonstationary states
611
Speech recognition using integra-normalizer and neuro-fuzzy method
612
Speech Recognition Using Linear Dynamic Models
613
Speech Recognition Using Long-Span Temporal Patterns in a Deep Network Model
614
Speech recognition using MFCC and DTW
615
Speech recognition using modified fuzzy hypersphere neural network
616
Speech recognition using multilayer perceptron
617
Speech recognition using multi-state acoustic and articulatory features models with asynchronous states transition
618
Speech Recognition Using Myoelectric Signals With Neural Networks
619
Speech recognition using neural networks
620
Speech Recognition Using Neural Networks
621
Speech recognition using neural networks
622
Speech recognition using neural networks with forward-backward probability generated targets
623
Speech recognition using noise-adaptive prototypes
624
Speech recognition using noise-adaptive prototypes
625
Speech recognition using peak frequency representations
626
Speech recognition using pulse coupled neural network
627
Speech recognition using Radial Basis Function neural network
628
Speech recognition using randomized relational decision trees
629
Speech recognition using reconstructed phase space features
630
Speech recognition using regularized minimum variance distortionless response spectrum estimation-based cepstral features
631
Speech recognition using segmental neural nets
632
Speech recognition using speaker adaptation by system parameter transformation
633
Speech recognition using stochastic segment neural networks
634
Speech recognition using sub-word neural tree network models and multiple classifier fusion
635
Speech recognition using sub-word units dependent on phonetic contexts of both training and recognition vocabularies
636
Speech recognition using syllable and pseudo articulatory features modeling
637
Speech Recognition Using Syllable Duration Ratio Model
638
Speech recognition using syllable-like units
639
Speech recognition using temporal decomposition and multi-layer feed-forward automata
640
Speech recognition using temporally connected kernels in mixture density hidden Markov models
641
Speech recognition using TES symbols
642
Speech recognition using the metric defined by integra-normalizer
643
Speech recognition using the modulation model
644
Speech recognition using three channel redundant wavelet filterbank
645
Speech recognition using time-warping neural networks
646
Speech recognition using variable frame rate coding
647
Speech recognition using voice-characteristic-dependent acoustic models
648
Speech recognition using wavelet packets, Neural Networks and Support Vector Machines
649
Speech recognition using weighted HMM and subspace projection approaches
650
Speech Recognition Using Zero-Crossing Features
651
Speech recognition using zero-crossing measurements and sequence information
652
Speech recognition via Hidden Markov Model and neural network trained by genetic algorithm
653
Speech recognition via STT API for autonomous mobile robot
654
Speech recognition with a competitive Probabilistic Radial Basis Neural Network
655
Speech recognition with a neural network trace-segmentation
656
Speech recognition with a noise-adapting codebook
657
Speech recognition with auxiliary information
658
Speech recognition with character string encoding
659
Speech recognition with continuous-parameter hidden Markov models
660
Speech recognition with deep recurrent neural networks
661
Speech Recognition With Flat Direct Models
662
Speech Recognition with Fuzzy Neural Network for Discrete Words
663
Speech recognition with high recognition rate by smoothed space pseudo Wigner-Ville distribution (SSPWD) and overlap slide window spectrum methods
664
Speech recognition with HMM models for cochlear prostheses
665
Speech recognition with large-scale speaker-class-based acoustic modeling
666
Speech Recognition with Limited Resources for Children and Adult Speakers
667
Speech recognition with localized time-frequency pattern detectors
668
Speech recognition with matrix-MCE based two-dimension-cepstrum in cars
669
Speech recognition with neural networks and network fusion
670
Speech recognition with prediction-adaptation-correction recurrent neural networks
671
Speech recognition with syllables and concepts
672
Speech recognition with the discrete rectangular wave transform
673
Speech recognition with very large size dictionary
674
Speech recognition with wavelet spectral subtraction in real noisy environment
675
Speech Recognition, Machine Translation, and Speech Translation—A Unified Discriminative Learning Paradigm [Lecture Notes]
676
Speech recognition: A current perspective: In spite of limitations, areas of application are growing, and voice communication with computers may well be commonplace by the 21st century
677
Speech recognition: A model and a program for research
678
Speech Recognition: A TutoriaI on the Subject
679
Speech Recognition: A Tutorial on the Subject
680
Speech Recognition: A Tutorial Overview
681
Speech recognition: Invited papers presented at the 1974 IEEE symposium
682
Speech recognition: theory and c++ implementation [Book Review]
683
Speech recognition´s evolution continues
684
Speech recognition-a time-frequency subspace filtering based approach
685
Speech Recognition-Based Mobile Geo-Mashup Application Technology
686
Speech recognition-can user problems be predicted?
687
Speech recognitionwith segmental conditional random fields: A summary of the JHU CLSP 2010 Summer Workshop
688
Speech recognitionwith speech density estimation by the Dirichlet Process Mixture
689
Speech recognizer for regional languages of Pakistan
690
Speech recognizer-based microphone array processing for robust hands-free speech recognition
691
Speech reconstruction for MFCC-based low bit-rate speech coding
692
Speech reconstruction from mel frequency cepstral coefficients and pitch frequency
693
Speech reconstruction from mel-frequency cepstral coefficients via ?1-norm minimization
694
Speech reconstruction via sparse representation using harmonic regularization
695
Speech reinforcement based on partial masking effect
696
Speech reinforcement in noisy reverberant conditions under an approximation of the short-time SII
697
Speech reinforcement in noisy reverberant environments using a perceptual distortion measure
698
Speech Reinforcement System for Car Cabin Communications
699
Speech reinforcement with a globally optimized perceptual distortion measure for noisy reverberant channels
700
Speech representation and transformation using adaptive interpolation of weighted spectrum: vocoder revisited
701
Speech representation based on instantaneous amplitudes
702
Speech research in telecommunications: a Bell-centric view
703
Speech resources for a Serbian LVCSR system
704
Speech restoration from LPC parameters corrupted by channel noise
705
Speech re-synthesis from spectrogram image through sinusoidal modelling
706
Speech resynthesis with a hardware synthesizer
707
Speech retrieval for TV news programs by fusing the audio and video information
708
Speech retrieval using spoken keyword extraction and semantic verification
709
Speech retrieval with video parsing for television news programs
710
Speech rhythm guided syllable nuclei detection
711
Speech rhythm-driven gesture animation
712
Speech sampling by level-crossing and its reconstruction using spline-based filtering
713
Speech Scenario Adaptation and Discourse Topic Recognition on Mobile Smart Terminal
714
Speech scrambling
715
Speech scrambling based on chaotic maps and one time pad
716
Speech scrambling by the matrixing of amplitude samples
717
Speech scrambling by the re-ordering of amplitude samples
718
Speech scrambling employing adaptive rate sampling
719
Speech scrambling prior to LPC coding
720
Speech scrambling with Hadamard transform in frequency domain
721
Speech segment selection for concatenative synthesis based on prosody-aligned distance measure
722
Speech segmentation and clustering based on speaker features
723
Speech segmentation and feature normalization based on area functions
724
Speech segmentation and recognition using adaptive linear prediction algorithm
725
Speech segmentation and recognition using syntactic methods on the direct signal
726
Speech segmentation and spoken document processing
727
Speech segmentation by variance fractal dimension
728
Speech segmentation in working environments
729
Speech segmentation using a hypothesis test based on Random Matrix Theory
730
Speech segmentation using divergence algorithm with Zero Crossing property
731
Speech Segmentation using Extrema-Based Signal Track Length Measure
732
Speech segmentation using multifractal measures and amplification of signal features
733
Speech segmentation using multilevel hybrid filters
734
Speech segmentation using probabilistic phonetic feature hierarchy and support vector machines
735
Speech segmentation without speech recognition
736
Speech segmentation without speech recognition
737
Speech segregation based on pitch tracking and amplitude modulation
738
Speech segregation based on sound localization
739
Speech Segregation Using an Auditory Vocoder With Event-Synchronous Enhancements
740
Speech segregation using event synchronous auditory vocoder
741
Speech selection and environmental adaptation for asynchronous speech recognition
742
Speech Sentence Compression Based on Speech Segment Extraction and Concatenation
743
Speech separation algorithms for multiple speaker environments
744
Speech separation based on improved deep neural networks with dual outputs of speech features for both target and interfering speakers
745
Speech separation based on semi-blind kurtosis maximization with magnitude and energy distance
746
Speech Separation based on signal-noise-dependent deep neural networks for robust speech recognition
747
Speech separation based on the images analysis method in CASA
748
Speech Separation Based on The Statistics of Binaural Auditory Features
749
Speech separation by efficient combinatorial decoding of speech mixtures
750
Speech separation by kurtosis maximization
751
Speech separation by means of stationary least-squares harmonic estimation
752
Speech separation by simulating the cocktail party effect with a neural network controlled Wiener filter
753
Speech Separation from Background of Music Based on Single-channel Recording
754
Speech separation in humans and machines
755
Speech Separation in Humans and Machines
756
Speech separation microphone array based on law of causality and frequency domain processing
757
Speech separation of a target speaker based on deep neural networks
758
Speech Separation of Multiple Moving Speakers Using Multisensor Multitarget Techniques
759
Speech Separation Using an Adaptive Sparse Dictionary Algorithm
760
Speech separation using DUET and binary masking with temporal smoothing in cepstral domain
761
Speech signal analysis and synthesis via Fourier-Bessel representation
762
Speech signal analysis using Fourier-Bessel Expansion and Hilbert Transform Separation Algorithm
763
Speech signal analysis using the Wigner distribution
764
Speech Signal Analysis with Exponential Autoregressive Model
765
Speech signal analysis with reallocated spectrogram
766
Speech signal band width extension and noise removal using subband HMN
767
Speech signal captured by PVDF sensor
768
Speech Signal Compression & DSP Realization with Virtual-Value Mapping Coding
769
Speech signal compression using sub-band division and sub-adaptive polynomial mapping quantization
770
Speech signal detection at the output of a cochlear model
771
Speech signal digital processing algorithm development environment
772
Speech Signal Enhancement Based on Adaptive Multi-resolution Form of SVD
773
Speech Signal Enhancement Based on MAP Algorithm in the ICA Space
774
Speech signal enhancement through wavelet domain MMSE filtering
775
Speech signal enhancement under multiple interferences using transfer function ratio beamformer
776
Speech signal enhancement using neural network and wavelet transform
777
Speech Signal Extraction Utilizing PCA-ICA Algorithm With a Non-Uniform Spacing Microphone Array
778
Speech Signal Feature Extraction Based on Wavelet Transform
779
Speech signal filtering by artificial neural networks
780
Speech signal improved transmission system for personal communications
781
Speech signal noise reduction by EMD
782
Speech signal prediction using feedforward neural network
783
Speech signal processing based emotion recognition
784
Speech signal processing basing upon delta modulation
785
Speech signal processing implementation for hearing impairment based on FPGA
786
Speech signal processing in order to increase recognition of spoken language
787
Speech signal processing using a time-frequency distribution of Cohen´s class
788
Speech signal processing using neural networks mapping the phonology of Sanskrit language using neural networks
789
Speech signal recognition based on genetic algorithm and Fisher projection
790
Speech signal reconstruction based on higher order spectra
791
Speech signal reconstruction based on the symmetric tight wavelet frame decomposition
792
Speech signal reconstruction for real-time voice transmission over the Internet
793
Speech signal recovery in colored noise using an adaptive Kalman filtering
794
Speech signal recovery in packet-switched communication networks
795
Speech signal recovery in packet-switched communication networks
796
Speech signal recovery in packet-switched communication networks
797
Speech signal recovery in white noise using an adaptive Kalman filter
798
Speech Signal Representation via Dictionary Learning in STFT Transformed Domain
799
Speech signal resampling by arbitrary rate
800
Speech signal restoration using an optimal neural network structure
801
Speech signals analysis using a frequency detector and smoothing first and second derivatives
802
Speech signals representation by Discrete Transforms
803
Speech signals separation: a new approach exploiting the coherence of audio and visual speech
804
Speech silicon AM: an FPGA-based acoustic modeling pipeline for hidden Markov model based speech recognition
805
Speech Sound Classification and Detection of Articulation Disorders with Support Vector Machines and Wavelets
806
Speech sounds and features
807
Speech sounds and features
808
Speech Source 3D Localization Focusing Algorithms Based on Microphone Array
809
Speech source inference
810
Speech source localization based on copula rank correlation matrix
811
Speech source localization in near field
812
Speech source localization using a multi-resolution technique
813
Speech source separation and noise reduction using a MMSE short-time spectral amplitude estimator
814
Speech Source Separation by Combining Localization Cues with Mixture Models of Speech Spectra
815
Speech Spectral Envelope Enhancement by HMM-Based Analysis/Resynthesis
816
Speech spectral envelope estimation through explicit control of peak evolution in time
817
Speech spectral segmentation for spectral estimation and formant modelling
818
Speech spectrogram based model adaptation for speaker identification
819
Speech spectrograms using the fast Fourier transform
820
Speech spectrum modeling applied to spectrum coding and prediction
821
Speech Spectrum Modeling for Joint Estimation of Spectral Envelope and Fundamental Frequency
822
Speech spectrum quantization using Gaussian mixture models and multi-dimensional companding
823
Speech spectrum representation and coding using multigrams with distance
824
Speech spectrum transformation by speaker interpolation
825
Speech steganalysis using chaotic-type features
826
Speech stream detection and enhancement wavelet-basedin high noise environment
827
Speech stream detection based on higher-order statistics
828
Speech stream detection based on one and half spectrum
829
Speech Stream Detection in Short-wave Channel Based on Cepstral of Auditory Wavelet Packets
830
Speech Stream Detection in Strong Noise based on Linear Prediction
831
Speech stuttering assessment using sample entropy and Least Square Support Vector Machine
832
Speech style conversion based on the statistics of vowel spectrograms and nonlinear frequency mapping
833
Speech summarization using clustered frames acoustic units for Persian news
834
Speech Summarization without Lexical Features for Mandarin Presentation Speech
835
Speech support in wireless, multihop networks
836
Speech synchronization between speech and lip shape movements for service robotics applications
837
Speech synthesis
838
Speech synthesis
839
Speech synthesis algorithm for Thai cochlear implants
840
Speech Synthesis Algorithm of Co-articulation Based on the Continuous Transition of Energy
841
Speech synthesis and the spectral estimation problem
842
Speech synthesis based on AMR-WB algorithm
843
Speech synthesis based on feature extraction to enhance noise-corrupted speech
844
Speech Synthesis Based on Hidden Markov Models
845
Speech synthesis based on PSOLA algorithm and modified pitch parameters
846
Speech synthesis by a programmable digital filter
847
Speech synthesis by analogy: recent advances and results
848
Speech synthesis by concatenating sub-syllabic sound units
849
Speech synthesis by dyads and automatic intonation processing
850
Speech synthesis by linear interpolation of spectral parameters between dyad boundaries
851
Speech synthesis by mimicking articulatory movements
852
Speech synthesis by rule using an optimal selection of non-uniform synthesis units
853
Speech synthesis for a specific speaker based on a labeled speech database
854
Speech synthesis for Bangla Text to Speech conversion
855
Speech synthesis for mobile phone
856
Speech synthesis for people with a visual impairment in digital television receivers
857
Speech synthesis for text-to-speech alignment and prosodic feature extraction
858
Speech synthesis from diphones
859
Speech synthesis from english text: A progress report
860
Speech synthesis from real time ultrasound images of the tongue
861
Speech synthesis from short-time Fourier transform magnitude and its application to speech processing
862
Speech synthesis from surface electromyogram signal
863
Speech synthesis from text
864
Speech synthesis from vocal tract area function acoustical measurements
865
Speech synthesis in a computer aided learning environment
866
Speech synthesis in augmentative communication
867
Speech synthesis in CAL/CBT
868
Speech synthesis in telecommunications
869
Speech synthesis in the time domain from text
870
Speech synthesis method based on application-specific synthesis units and its implementation on a 32-bit microprocessor
871
Speech synthesis method based on application-specific synthetic units and its implementation on a 32-bit microprocessor
872
Speech synthesis method with a harmonic plus noise model
873
Speech synthesis models: a review
874
Speech synthesis module for Romanian language
875
Speech synthesis of phonemic triplets through a neural network-controlled formant synthesizer
876
Speech Synthesis Research Based on EGG
877
Speech Synthesis Software For A 32-bit Microprocessor
878
Speech synthesis software for a 32-bit microprocessor
879
Speech synthesis software with a variable speaking rate and its implementation on a 32-bit microprocessor
880
Speech synthesis software with variable speaking rate and its implementation on a 32-bit microprocessor
881
Speech synthesis system based on a variable decimation/interpolation factor
882
Speech synthesis techniques
883
Speech synthesis techniques. A survey
884
Speech synthesis using a nonlinear energy damping model for the vocal folds vibration effect
885
Speech synthesis using approximate matching of syllables
886
Speech synthesis using articulatory-knowledge based HMM structure
887
Speech synthesis using artificial neural networks
888
Speech synthesis using compressed database
889
Speech synthesis using HMM based diphone inventory encoding for low-resource devices
890
Speech synthesis using HMMs with dynamic features
891
Speech synthesis using real-time software
892
Speech synthesis using stochastic Markov graphs
893
Speech synthesis using subband-coded multiband source components and sinusoids
894
Speech synthesis using the CELP algorithm
895
Speech synthesis using two-sided linear prediction parameters
896
Speech synthesis using warped linear prediction and neural networks
897
Speech synthesis with artificial neural networks
898
Speech synthesis: A technology that speaks for itself: Each method has its own trade-off. High quality output limits your vocabulary, while a more mechanical sound lets you say more
899
Speech synthesis: Design case history: Speak & Spell learns to talk: Designers of an electronic learning aid mastered advanced speech technology and meticulously `prepared¿ the consumer
900
Speech synthesizer with speculative multithreading and speculative computation reuse
901
Speech systems classification based on frequency of binary word features
902
Speech Tagging Based Improvement of the RSS Polymerization News
903
Speech Technical Committee
904
Speech technology and applications in the intelligent network
905
Speech technology and computing: a unique partnership
906
Speech technology and information access [In the Spotlight]
907
Speech technology and systems in human-machine communication [from the Guest Editors]
908
Speech technology applications to telephone services
909
Speech technology for multimedia content management
910
Speech technology helps the police with their inquiries
911
Speech technology opportunities and challenges
912
Speech technology: a solution for people with disabilities
913
Speech Text-Independent Segmentation Using an Improvement Method for Identification of Phoneme Boundaries
914
Speech Therapy and Assessment (via Multimedia Devices for Cleft Lip and Palate Patients)
915
Speech therapy software on an open web platform
916
Speech therapy using computer based minimal consonant pair discrimination
917
Speech timing and cross-linguistic studies towards computational human modeling
918
Speech timing of coreference
919
Speech to facial animation conversion for deaf customers
920
Speech to text and accelerometer based smart phone interaction system
921
Speech traffic capacity of idle-signal casting multiple access with collision detection (ICMA/CD) for cellular mobile radio
922
Speech training devices for profoundly deaf children
923
Speech training system based on resonant frequencies of vocal tract
924
Speech training systems for handicapped children using vocal tract lateral shapes
925
Speech trajectory discrimination using the minimum classification error learning
926
Speech trajectory recognition in SOFM by using Bayes theorem
927
Speech transcoding from AMR to G.729 in excitation domain
928
Speech transcript analysis for automatic search
929
Speech transcription in multiple languages
930
Speech transformations based on a sinusoidal representation
931
Speech transformations based on a sinusoidal representation
932
Speech Translation by Confusion Network Decoding
933
Speech Translation Enhanced ASR for European Parliament Speeches - On the Influence of ASR Performance on Speech Translation
934
Speech translation enhanced automatic speech recognition
935
Speech translation on Internet CEST-CAS2.0
936
Speech Translation Statistical System Using Multimodal Sources of Knowledge
937
Speech Translation with Phrase Based Stochastic Finite-State Transducers
938
Speech translation: coupling of recognition and translation
939
Speech transmission capacity of standard ethernet systems
940
Speech transmission in TETRA [private mobile radio]
941
Speech transmission performance planning in hybrid IP/SCN networks
942
Speech Transmission Using an Adaptive Burst Mode Technique
943
Speech transmission using DPCM coding, partial response continuous phase modulation and diversity
944
Speech transmission using rate-compatible trellis codes and embedded source coding
945
Speech Transport in Packet-Radio Networks with Mobile Nodes
946
Speech under physical stress: A production-based framework
947
Speech under stress conditions: overview of the effect on speech production and on system performance
948
Speech understanding and spoken dialogue systems
949
Speech understanding in noise provided by a simulated cochlear implant processor based on matching pursuit
950
Speech Understanding Performance of Cochlear Implant Subjects Using Time–Frequency Masking-Based Noise Reduction
951
Speech understanding system using knowledge engineering techniques
952
Speech understanding systems
953
Speech Understanding Through Syntactic and Semantic Analysis
954
Speech understanding using a unification grammar
955
Speech Unit Selection Based on Matching Pursuit
956
Speech unit selection based on target values driven by speech data in concatenative speech synthesis
957
Speech user interface for computer based education system
958
Speech utterance categorisation given one training utterance per category
959
Speech utterance classification
960
Speech Utterance Classification Model Training without Manual Transcriptions
961
Speech variability compensation for expressive speech synthesis
962
Speech variability in automatic speaker recognition systems for commercial and forensic purposes
963
Speech variability in automatic speaker recognition systems for forensic purposes
964
Speech visualization based on Auditory Model for hearing impaired
965
Speech visualization based on improved spectrum for deaf children
966
Speech Visualization based on Locally Linear Embedding (LLE) for the Hearing Impaired
967
Speech Visualization based on Robust Self-organizing Map (RSOM) for the Hearing Impaired
968
Speech visualization based on wavelet transform for the hearing impaired
969
Speech visualization by integrating features for the hearing impaired
970
Speech Visualization for Learning Pronunciation Based on Auditory Characteristics
971
Speech visualization of Chinese Mandarin single-finals
972
Speech visualization of some Mandarin compound-finals
973
Speech visualization research based on combined feature and neural network
974
Speech visualization simulation research for deaf-mute
975
Speech visualization system based on integrating feature for the hearing handicapped
976
Speech volumes on bell system message circuits
977
Speech vs music discrimination using Empirical Mode Decomposition
978
Speech vs Nonspeech Segmentation of Audio Signals Using Support Vector Machines
979
Speech vs. touch-tone: Telephony interfaces for information access by low literate users
980
Speech VTF similarity property and its application on coding analysis
981
Speech Watermarking Based on Wavelet Transform and BCH Coding
982
Speech watermarking for air traffic control
983
Speech Watermarking for Analog Flat-Fading Bandpass Channels
984
Speech watermarking with objective fidelity and robustness criteria
985
Speech Waveform Compression Using Robust Adaptive Voice Activity Detection for Nonstationary Noise in Multimedia Communications
986
Speech zero-crossing rate compression for bandwidth compression
987
Speechߞman´s natural communication
988
Speech, audio, and acoustic processing for multimedia
989
Speech, data, and channel coding for digital cellular mobile
990
Speech, Emotion, Age, Language, Task, and Typicality: Trying to Disentangle Performance and Feature Relevance
991
Speech, image and video processing
992
Speech, Image and Video Processing
993
Speech. Coding using Entropy-Constrained Code-Excited Linear Predictive Quantization (EC-CELP)
994
Speech/Audio Signal Classification Using Spectral Flux Pattern Recognition
995
Speech/gesture interface to a visual computing environment for molecular biologists
996
Speech/gesture interface to a visual-computing environment
997
Speech/Music Classification of Short Audio Segments
998
Speech/Music Classification Using Empirical Mode Decomposition
999
Speech/Music Classification Using Occurrence Pattern of ZCR and STE
1000
Speech/Music Discrimination Based on Spectral Peak Analysis and Multi-layer Perceptron
