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The paper presents an implementation of an improved smartphone-based hearing aid which originates from our free smartphone application “Petralex” released for iOS and Android devices. In the present contribution we develop a new processing scheme that decomposes the signal into perceptually matched sliding bands and implements combined noise reduction and acoustic feedback suppression (AFS). The proposed...
The demonstration presents a real-time mockup of smartphone- based hearing aid with combined noise and acoustic feedback reduction. The designed reduction algorithm is based on spectral weighting approach which makes it very robust to rapid changes in feedback path either caused by displacement of the speaker/microphone or room acoustics. The aim of the demonstration is to show potential of the implemented...
The paper presents an algorithm for accurate pitch estimation that takes advantage of the sinusoidal model with instantaneous parameters. The algorithm decomposes the signal into subband components, extracts their instantaneous parameters and evaluates period candidate generating function (PCGF). In order to achieve high accuracy for low and high-pitched sounds it is assumed that possible pitch variation...
The paper presents an approach to subchannel decomposition of multidimensional signals that is performed by means of a DFT-modulated filter bank designed for multidimensional signals. An r-dimension signal can be decomposed using r scalar DFT-modulated filter banks. However, the proposed vectorized filter bank allows to obtain polyphase filter bank structure that can be efficiently implemented in...
The paper presents a solution for singing voice processing that is used in a karaoke application with automated voice correction1. The intended purpose of the application is to automatically improve user's performance towards performance of a professional singer by implementation of voice effects such as pitch correction, artificial polyphony, time stretching and other. The proposed framework incorporates...
The paper addresses the problem of estimation of amplitude envelope and instantaneous frequency of an amplitude and frequency modulated (AM-FM) signal in noisy conditions. The algorithm proposed in the paper utilizes derivatives of the signal and is analogous to well-known Energy Separation Algorithms (ESA) based on Teager-Kaiser energy operator (TEO). The formulation of the algorithm is based on...
A family of multiplierless transforms is presented that approximate the eight-point type-II discrete cosine transform (DCT) as accurately as the state-of-the-art scaled DCT schemes, but having 14??17% shorter critical paths (1/6 or 1/7 less adders). Compared to the existing solutions that use the coordinate rotation digital computer (CORDIC) algorithm, the advantage of higher throughput is accompanied...
The paper presents a systematic approach to synthesizing and implementing fast algorithms for computing the DCT of a power-of-two size. The main features of the obtainable algorithms are regularity of their signal flow graphs and low arithmetic complexity. Multiplierless implementation of the algorithms is based on the algebraic integer (AI) technique. A general AI encoding scheme for fast DCT algorithms...
The paper presents a pitch estimation technique based on the robust algorithm for pitch tracking (RAPT) framework. The proposed solution provides estimation of instantaneous pitch values and is not sensitive to rapid frequency modulations. The technique utilizes a different period candidate generating function based on instantaneous harmonic parameters. The function is similar to normalized cross-correlation...
The paper presents a hearing aid (HA) system based on a low-delay cochlear filter bank derived from the discrete cochlear model. The spectral gain shaping method (SGSM) that includes dynamic range compression, hearing loss compensation and noise reduction is applied in perceptually matched frequency bands. The acoustic feedback cancellation is implemented as an off-the-forward path scheme and does...
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