It is generally desirable in music reproduction but can be detrimental to speech-related applications. For the human listener, while the early reflections help to improve speech intelligibility, the late reflections have been shown to impair perceived speech quality. For speech processing technologies such as automatic speech recognizers, reverberation reduces accuracy and performance. Dereverberation is therefore an important research topic with interest driven by increasing availability of communication devices and consumer demand.
One approach to dereverberation computes a set of equalizing filters that are used to perform the dereverberation processing, given multichannel inputs and estimates of the acoustic impulse responses AIRs between the source signal and microphones. However, estimation errors are inevitable in practice and therefore robust channel equalizers are required. This thesis aims to develop such robust algorithms in a manner that is desirable specifically for speech dereverberation.
The framework of channel shortening is used, having been previously shown to give promising results. For the speech enhancement task, the proposed dynamic feature constraint help to improve cepstral distance, frequency-weighted segmental signal-to-noise ratio SNR , and log likelihood ratio metrics while moderately degrades the speech-to-reverberation modulation energy ratio. In addition, the cross transform feature adaptation improves the ASR performance significantly for clean-condition trained acoustic models.
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 Speech Dereverberation with Context-aware Recurrent Neural Networks
How to get this document? There is an Open Access version for this licensed article that can be read free of charge and without license restrictions. The content of the Open Access version may differ from that of the licensed version. Document information Title:.
- Blind channel identification for speech dereverberation using l1-norm sparse learning;
- Buddhist Translation: Problems and Perspectives.
- Regularization for Partial Multichannel Equalization for Speech Dereverberation!
- GitHub - helianvine/fdndlp: A speech dereverberation algorithm, also called wpe.
Speech dereverberation for enhancement and recognition using dynamic features constrained deep neural networks and feature adaptation. Beamforming , Deep neural networks , Dynamic features , Feature adaptation , Robust speech recognition , Reverberation challenge , Speech enhancement.
Table of contents — Volume , Issue 1 Show all volumes and issues Tables of content are generated automatically and are based on records of articles contained that are available in the TIB-Portal index. Resolution enhancement for ISAR imaging via improved statistical compressive sensing. Sparsity-aware multiple relay selection in large multi-hop decode-and-forward relay networks.
Class-specific discriminant time-frequency analysis using novel jointly learnt non-negative matrix factorization. The generalized frequency-domain adaptive filtering algorithm as an approximation of the block recursive least-squares algorithm. Physics-driven quantized consensus for distributed diffusion source estimation using sensor networks. Smoothing data association for target trajectory estimation in cluttered environments. Advances in adaptive filtering theory and applications to acoustic and speech signal processing. Fault detection reduced-order filter design for discrete-time Markov jump system with deficient transition information.
Range cell migration correction analysis of one-step and two-step motion compensation for millimeter-wave airborne SAR imaging. Exploiting sensor mobility and covariance sparsity for distributed tracking of multiple sparse targets. Efficiently sphere-decodable physical layer transmission schemes for wireless storage networks. High-order balanced M-band multiwavelet packet transform-based remote sensing image denoising.
Singular spectrum-based matrix completion for time series recovery and prediction. Joint training of DNNs by incorporating an explicit dereverberation structure for distant speech recognition.
A recursive kinematic random forest and alpha beta filter classifier for 2D radar tracks. Nonlinear speech enhancement by virtual increase of channels and maximum SNR beamformer. Distributed Gram-Schmidt orthogonalization with simultaneous elements refinement. A limited feedback scheme for massive MIMO systems based on principal component analysis.
Practical security and privacy attacks against biometric hashing using sparse recovery. Reduced-dimension space-time adaptive processing based on angle-Doppler correlation coefficient.
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Optimization of a MIMO amplify-and- forward relay system with channel state information estimation error and feedback delay. Range-azimuth decouple beamforming for frequency diverse array with Costas-sequence modulated frequency offsets. A hybrid passive localization method under strong interference with a preliminary experimental demonstration. Power-efficient distributed resource allocation under goodput QoS constraints for heterogeneous networks.
Optimal fusion rule for distributed detection in clustered wireless sensor networks.
Class-specific kernel linear regression classification for face recognition under low-resolution and illumination variation conditions. Efficient non-uniform deblurring based on generalized additive convolution model. A novel sequential algorithm for clutter and direct signal cancellation in passive bistatic radars.
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Low probability of intercept-based adaptive radar waveform optimization in signal-dependent clutter for joint radar and cellular communication systems. Unimodular sequence design under frequency hopping communication compatibility requirements. Autonomous radar pulse modulation classification using modulation components analysis.
Identification of piecewise affine systems based on fuzzy PCA-guided robust clustering technique.fr.ziwusymowoji.tk
Speech Dereverberation Using Linear Prediction
Online rate adjustment for adaptive random access compressed sensing of time-varying fields. A summary of the REVERB challenge: state-of-the-art and remaining challenges in reverberant speech processing research. Performance analysis of transmit antenna selection for cognitive radio systems with imperfect channel estimation.
Color interpolation algorithm for an RWB color filter array including double-exposed white channel. Robust adaptive filtering using recursive weighted least squares with combined scale and variable forgetting factors.
A novel resource scheduling method of netted radars based on Markov decision process during target tracking in clutter. Automatic and online setting of similarity thresholds in content-based visual information retrieval problems. Matched, mismatched, and robust scatter matrix estimation and hypothesis testing in complex t-distributed data. Rate-distortion-optimized multi-view streaming in wireless environment using network coding.
Multicast Wi-Fi Raptor-enabled data carousel design: simulation and practical implementation. A residual range cell migration correction algorithm for bistatic forward-looking SAR. OFDMA-based network-coded cooperation: design and implementation using software-defined radio nodes.
Robust group compressive sensing for DOA estimation with partially distorted observations.