Aug 2026· International journal of computer information systems and industrial management applications· 0 citations
TL;DR
Experimental results indicate that CSM accelerates the training of the models while enhancing the speech quality and show that the method effectively denoises noisy speech audio while being both efficient and accurate.
Abstract
Noise suppression that improves speech intelligibility is critical in teleconferencing, Voice over Internet Protocol (VoIP), hearing aids, and speech recognition systems. This work has attempted an adaptive filtering technique for noise suppression to improve the performance of speech signals. In this regard, two of the most popular algorithms, namely the Recursive Least Squares (RLS) algorithm and Normalised Least Mean Squares (NLMS) algorithm, specifically for existing speech distortion enhancement by room noise, will be analysed. This way, such adaptive filtering techniques can help mitigate those disturbances and enhance speech intelligibility and sound reproduction or processing performance. Phase enhancement to improve the perceptual quality of synthesised speech has recently attracted considerable attention among speech researchers. A few researchers have directly integrated phase estimation modules into speech enhancement architecture with complex-valued time-frequency (T-F) domain representations, such as the complex ratio mask (CRM), for which the phase can be computed from the real and imaginary parts of the complex STFT spectrogram. Unfortunately, spectrogram masking violates these consistency constraints, which can lead to artefacts in the extracted component while also unnecessarily widening the solution space. To address this issue, we propose consistency-based constraints with a novel spectrogram permutation masking technique called Consistency Spectrogram Masking (CSM) to improve the complex spectrogram estimation. Experimental results indicate that CSM accelerates the training of the models while enhancing the speech quality. Our experimental results counter this and show that the method effectively denoises noisy speech audio while being both efficient and accurate.
Noise reduction is crucial to improving real-time acoustic speech communication quality and intelligibility. We present a low-complexity adaptive noise cancellation system based on the Least Mean Squares (LMS) algorithm in a Digital Signal Processing (DSP) environment. The main objective of this research is to reduce t...
Mahesh Mudavath, Chiyttaveni Komaltha, K. S. Babu et al.· International journal of com...· 0 citations
Environmental noise can substantially affect speech quality and intelligibility in real-time audio systems, particularly when speech enhancement is implemented on computationally limited edge platforms. Conventional frequency-domain techniques, including spectral subtraction and Wiener filtering, provide computationall...
Sherin Thomas, D. Sugumar· International Conference Inn...· 0 citations
Overall, this review provides a unified perspective on speech enhancement techniques, identifies their strengths, and highlights emerging research opportunities for the development of robust, efficient, and intelligent speech enhancement systems.
Pushpraj Tanwar, A. Somkuwar, Rakesh Kumar Gumasta· Engineer· 0 citations
In practical voice communication and processing, speech signals are extremely sensitive to background noise, equipment noise, and interference from complex environments, leading to a decline in speech quality and clarity. This, in turn, affects the overall performance of downstream systems such as speech recognition, s...
Wen Fan, Wei-Yu Liang, Duo-Duo Han et al.· 電腦學刊· 0 citations
Mask-based speech enhancement is widely used for suppressing noise and interference, but its performance in spatial audio algorithms with multichannel output has not been studied extensively. In such settings, speech enhancement must improve speech quality while preserving spatial cues that are essential for localizati...
Sheli Hendel, B. Rafaely, Dorothea Kolossa· 0 citations
Active noise cancellation (ANC) is common in headphones but rare in hearing aids, and its value for understanding speech in noise is not established. Algorithms that improve signal-to-noise ratio (SNR) act only on the hearing aid's amplified path, while unprocessed sound also reaches the eardrum directly through the ve...
Andrew T. Sabin, Lin-Kai Li· 0 citations
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