An Efficient Reference Free Adaptive Learning Process for Speech Enhancement Applications
Source abstract
Abstract
In issues like hearing impairment, speech therapy and hearing aids play a major role in reducing the impairment. Removal of noise signals from speech signals is a key task in hearing aids as well as in speech therapy. During the transmission of speech signals, several noise components contaminate the actual speech components. This paper addresses a new adaptive speech enhancement (ASE) method based on a modified version of singular spectrum analysis (MSSA). The MSSA generates a reference signal for ASE and makes the ASE is free from feeding reference component. The MSSA adopts three key steps for generating the reference from the contaminated speech only. These are decomposition, grouping and reconstruction. The generated reference is taken as a reference for variable size adaptive learning algorithms. In this work two categories of adaptive learning algorithms are used. They are step variable adaptive learning (SVAL) algorithm and time variable step size adaptive learning (TVAL). Further, sign regressor function is applied to adaptive learning algorithms to reduce the computational complexity of the proposed adaptive learning algorithms. The performance measures of the proposed schemes are calculated in terms of signal to noise ratio improvement (SNRI), excess mean square error (EMSE) and misadjustment (MSD). For cockpit noise these measures are found to be 29.2850, –27.6060 and 0.0758 dB respectively during the experiments using SVAL algorithm. By considering the reduced number of multiplications the sign regressor version of SVAL based ASE method is found to better then the counter parts.
Source: openalex_inverted_index · Quality: baseline
Research context
OpenAlex topics: Speech and Audio Processing · Advanced Adaptive Filtering Techniques · Blind Source Separation Techniques
SciVX discovery tags are terminology-based. OpenAlex topics are source-provided classifications.
References (29)
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1. A Bayesian hierarchical model for speech enhancement with time-varying audio channel10.1109/TASLP.2018.2876177 ↗
2. A spectra-based equalization-generation combined framework for throat microphone speech enhancement10.1109/ACCESS.2018.2879689 ↗
3. Adaptive multi-band filter structure-based far-end speech enhancement10.1049/iet-spr.2019.0226 ↗
4. Adaptive time segmentation for improved speech enhancement10.1109/TASL.2006.872596 ↗
5. Audio-visual speech enhancement using multimodal deep convolutional neural networks10.1109/TETCI.2017.2784878 ↗
6. Compressive sensing-based speech enhancement10.1109/TASLP.2016.2598306 ↗
7. Noise reduction in industry based on virtual instrumentation10.32604/cmc.2021.017568 ↗
8. Applying and comparison of chaotic-based permutation algorithms for audio encryption10.32604/cmc.2021.015175 ↗
9. Deep neural network-based linear predictive parameter estimations for speech enhancement10.1049/iet-spr.2016.0477 ↗
10. Hidden Markov model-based speech enhancement using multivariate Laplace and Gaussian distributions10.1049/iet-spr.2014.0032 ↗
11. Robust speaker recognition based on single-channel and multi-channel speech enhancement10.1109/TASLP.2020.2986896 ↗
12. Low-distortion MMSE speech enhancement estimator based on Laplacian prior10.1109/ACCESS.2017.2699782 ↗
13. Multi-level single-channel speech enhancement using a unified framework for estimating magnitude and phase spectra10.1109/TASLP.2020.2986877 ↗
14. New results in modulation-domain single-channel speech enhancement10.1109/TASLP.2017.2747082 ↗
15. Performance improvement of adaptive wavelet thresholding for speech enhancement using generalized Gaussian priors and frame-wise context modeling10.1109/ACCESS.2020.3023348 ↗
16. Phase-based dual-microphone speech enhancement using a prior speech model10.1109/TASL.2006.876870 ↗
17. Semi-supervised multichannel speech enhancement with a deep speech prior10.1109/TASLP.2019.2944348 ↗
18. Single-channel speech enhancement based on adaptive low-rank matrix decomposition10.1109/ACCESS.2020.2975069 ↗
19. Speech enhancement based on Bayesian low-rank and sparse decomposition of multichannel magnitude spectrograms10.1109/TASLP.2017.2772340 ↗
20. Speech enhancement based on full-sentence correlation and clean speech recognition10.1109/TASLP.2017.2651406 ↗
21. Adaptive noise cancellers for cardiac signal enhancement for IOT based health care systems
22. Efficient adaptive noise cancellation techniques in an IOT enabled telecardiology system10.14419/ijet.v7i2.17.11562 ↗
23. P. Srinivasareddy and M. Z. Ur Rahman, “Novel simplified logarithmic adaptive exon prediction for DNA analysis,” vol. 10, pp. 1422–1432, 2018.
24. Novel logarithmic reference free adaptive signal enhancers for ECG analysis of wireless cardiac care monitoring systems10.1109/ACCESS.2018.2866303 ↗
25. Cloud-based adaptive exon prediction for DNA analysis10.1049/htl.2017.0032 ↗
26. Motion artifact removal from single channel electroencephalogram signals using singular spectrum analysis10.1016/j.bspc.2016.06.017 ↗
27. Haykin · Adaptive Filter Theory · 2007
28. A novel approach for motion artifact reduction in PPG signals based on AS-LMS adaptive filter10.1109/TIM.2011.2175832 ↗
29. An efficient Kalman noise canceller for cardiac signal analysis in modern telecardiology systems10.1109/ACCESS.2018.2848201 ↗