Alexander A. Petrovsky

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This paper addresses the problem of noise estimation for the Karhunen-Loeve transform (KLT) based speech enhancement. The eigenvalues and eigenvectors of the noise covari-ance matrix are tracked using recursive averaging algorithm. This process is controlled by the noise power minima obtained from the noisy signal even during the speech activity periods.(More)
This paper introduces a framework for parametric speech modeling that can be used in various speech applications such as text-to-speech synthesis, voice conversion etc. In order to reduce impact of pitch variations the harmonic analysis is done in the warped time scale that is aligned with instantaneous pitch values. It is assumed that each harmonic has its(More)
A novel approach to the design and implementation of four-channel paraunitary filter banks is presented. It utilizes hypercomplex number theory, which has not yet been employed in these areas. Namely, quaternion multipliers are presented as alternative pa-raunitary building blocks, which can be regarded as generalizations of Givens (planar) rotations. The(More)
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(More)
Speech recognition engines should remain reasonably accurate in adverse environments in order to find their ways from laboratories towards applications. However the human auditory system has been proven to be a versatile tool, which is capable of outperforming the known artificial algorithms in their target environments. Recent advances in psychoacoustics(More)
*) This paper presents an alternative factorization for 8-channel general paraunitary filter bank. The utilization of quaternion multiplications leads to a lattice structure being lossless regardless of coefficient quantization. Other advantages are reduced memory requirements and good suitability for FPGA and VLSI implementations. The shown decompositions(More)
This paper describes a method of accurate estimation of the instantaneous speech signal harmonic parameters. The method is based on adaptive filtering of the speech signal along its harmonic components. A simple way of filter synthesis based on the Fourier transform is also proposed. The synthesized filters have a closed form impulse response which can be(More)