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"This book provides various speech enhancement algorithms for digital hearing aids. It covers information on noise signals extracted from silences of speech signal. The description of the algorithm used for this purpose is also provided. Different types of adaptive filters such as Least Mean Squares (LMS), Normalized LMS (NLMS) and Recursive Lease Squares (RLS) are described for noise reduction in the speech signals. Different types of noises are taken to generate noisy speech signals, and therefore information on various noises signals is provided. The comparative performance of various adaptive filters for noise reduction in speech signals is also described. In addition, the book provides a speech enhancement technique using adaptive filtering and necessary frequency strength enhancement using wavelet transform as per the requirement of audiogram for digital hearing aids.- Presents speech enhancement techniques for improving performance of digital hearing aids;- Covers various types of adaptive filters and their advantages and limitations;- Provides a hybrid speech enhancement technique using wavelet transform and adaptive filters."

"This book provides various speech enhancement algorithms for digital hearing aids. It covers information on noise signals extracted from silences of speech signal. The description of the algorithm used for this purpose is also provided. Different types of adaptive filters such as Least Mean Squares (LMS), Normalized LMS (NLMS) and Recursive Lease Squares (RLS) are described for noise reduction in the speech signals. Different types of noises are taken to generate noisy speech signals, and therefore information on various noises signals is provided. The comparative performance of various adaptive filters for noise reduction in speech signals is also described. In addition, the book provides a speech enhancement technique using adaptive filtering and necessary frequency strength enhancement using wavelet transform as per the requirement of audiogram for digital hearing aids.- Presents speech enhancement techniques for improving performance of digital hearing aids;- Covers various types of adaptive filters and their advantages and limitations;- Provides a hybrid speech enhancement technique using wavelet transform and adaptive filters."

"This book presents a statistical parametric speech synthesis (SPSS) framework for developing a speech synthesis system where the desired speech is generated from the parameters of vocal tract and excitation source. Throughout the book, the authors discuss novel source modeling techniques to enhance the naturalness and overall intelligibility of the SPSS system. This book provides several important methods and models for generating the excitation source parameters for enhancing the overall quality of synthesized speech. The contents of the book are useful for both researchers and system developers. For researchers, the book is useful for knowing the current state-of-the-art excitation source models for SPSS and further refining the source models to incorporate the realistic semantics present in the text. For system developers, the book is useful to integrate the sophisticated excitation source models mentioned to the latest models of mobile/smart phones.- Presents the efficient excitation source modeling techniques for generating high quality speech;- Includes a combination of both waveform and parametric methods to enhance the quality of synthesis;- Features and methods that need less memory and computational requirements than others, allowing them to be integrated to smart phones and smaller devices."

"This work addresses this problem in the short-time Fourier transform (STFT) domain. We divide the general problem into five basic categories depending on the number of microphones being used and whether the interframe or interband correlation is considered. The first category deals with the single-channel problem where STFT coefficients at different frames and frequency bands are assumed to be independent. The second category also concerns the single-channel problem. The difference is that now the interframe correlation is taken into account and a filter is applied in each subband instead of just a gain. The third and fourth classes discuss the problem of multichannel noise reduction in the STFT domain with and without interframe correlation, respectively. In the last category, consider the interband correlation in the design of the noise reduction filters. Illustrate the basic principle for the single-channel case as an example, while this concept can be generalized to other scenarios. In all categories, propose different optimization cost functions from which derive the optimal filters and we also define the performance measures that help analyzing them."

"This book covers newly developed and novel Steganography techniques and algorithms. The book outlines techniques to provide security to a variety of applications using Steganography, with the goal of both hindering an adversary from decoding a hidden message, and also preventing an adversary from suspecting the existence of covert communications. The book looks into applying these newly designed and improved algorithms to provide a new and efficient Steganographic system, called Characteristic Region-Based Image Steganography (CR-BIS). The algorithms combine both the robustness of the Speeded-Up Robust Features technique (SURF) and Discrete Wavelet Transform (DWT) to achieve characteristic region Steganography synchronization. The book also touches on how to avoid hiding data in the whole image by dynamically selecting characteristic regions for the process of embedding. - Applies and discusses innovative techniques for hiding text in a digital image file or even using it as a key to the encryption;- Provides a variety of methods to achieve characteristic region Steganography synchronization;- Shows how Steganography improves upon cryptography by using obscurity features."

"Novel techniques for dialectal Arabic speech describes approaches to improve automatic speech recognition for dialectal Arabic. Since speech resources for dialectal Arabic speech recognition are very sparse, the authors describe how existing Modern Standard Arabic (MSA) speech data can be applied to dialectal Arabic speech recognition, while assuming that MSA is always a second language for all Arabic speakers. "

"This book presents techniques for audio search, aimed to retrieve information from massive speech databases by using audio query words. The authors examine different features, techniques and evaluation measures attempted by researchers around the world. The topics covered also include available databases, software / tools, patents / copyrights, and different platforms for benchmarking. The content is relevant for developers, academics, and students.​"