2,244 research outputs found
Single-Channel Speech Enhancement Based on Frequency Domain ALE
In the present paper, a new single-channel speech enhancement system is proposed. The proposed system is based on frequency domain adaptive line enhancer; therefore, it is advantageous to non-stationary environments. Also, frequency domain decorrelation parameters are introduced and then adjusted independently. The performance of the proposed system is examined through computer simulations. The effectiveness of the proposed system is confirmed through computer simulations
Uncertainty Estimation in Deep Speech Enhancement Using Complex Gaussian Mixture Models
Single-channel deep speech enhancement approaches often estimate a single
multiplicative mask to extract clean speech without a measure of its accuracy.
Instead, in this work, we propose to quantify the uncertainty associated with
clean speech estimates in neural network-based speech enhancement. Predictive
uncertainty is typically categorized into aleatoric uncertainty and epistemic
uncertainty. The former accounts for the inherent uncertainty in data and the
latter corresponds to the model uncertainty. Aiming for robust clean speech
estimation and efficient predictive uncertainty quantification, we propose to
integrate statistical complex Gaussian mixture models (CGMMs) into a deep
speech enhancement framework. More specifically, we model the dependency
between input and output stochastically by means of a conditional probability
density and train a neural network to map the noisy input to the full posterior
distribution of clean speech, modeled as a mixture of multiple complex Gaussian
components. Experimental results on different datasets show that the proposed
algorithm effectively captures predictive uncertainty and that combining
powerful statistical models and deep learning also delivers a superior speech
enhancement performance.Comment: 5 pages, 4 figure
A Subband Hybrid Beamforming for In-car Speech Enhancement
Publication in the conference proceedings of EUSIPCO, Bucharest, Romania, 201
Theory, design and application of gradient adaptive lattice filters
SIGLELD:D48933/84 / BLDSC - British Library Document Supply CentreGBUnited Kingdo
Implementation and evaluation of a low complexity microphone array for speaker recognition
Includes bibliographical references (leaves 83-86).This thesis discusses the application of a microphone array employing a noise canceling beamforming technique for improving the robustness of speaker recognition systems in a diffuse noise field
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