15 research outputs found

    Vocal Processing with Spectral Analysis

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    A well-known signal processing issue is that of the “cocktail party problem,” which A well-known signal processing issue is that of the “cocktail party problem,” which refers to the need to be able to separate speakers from a mixture of voices. A solution to this problem could provide insight into signal separation in a variety of signal processing fields. In this study, a method of vocal signal processing was examined to determine if principal component analysis of spectral data could be used to characterize differences between speakers and if these differences could be used to separate mixtures of vocal signals. Processing was done on a set of voice recordings from thirty different speakers to create a projection matrix that could be used by an algorithm to identify the source of an unknown recording from one of the thirty speakers. Two different identification algorithms were tested. The first had an average correct prediction rate of 15.69%, while the second had an average correct prediction rate of 10.47%. Additionally, one principal component derived from the processing provided a notable distinction between principal values for male and female speakers. Males tended to produce positive principal values, while females tended to produce negative values. The success of the algorithm could be improved by implementing differentiation between time segments of speech and segments of silence. The incorporation of this distinction into the signal processing method was recommended as a topic for future study

    Sequential blind source separation based exclusively on second-order statistics developed for a class of periodic signals

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    A sequential algorithm for the blind separation of a class of periodic source signals is introduced in this paper. The algorithm is based only on second-order statistical information and exploits the assumption that the source signals have distinct periods. Separation is performed by sequentially converging to a solution which in effect diagonalizes the output covariance matrix constructed at a lag corresponding to the fundamental period of the source we select, the one with the smallest period. Simulation results for synthetic signals and real electrocardiogram recordings show that the proposed algorithm has the ability to restore statistical independence, and its performance is comparable to that of the equivariant adaptive source separation (EASI) algorithm, a benchmark high-order statistics-based sequential algorithm with similar computational complexity. The proposed algorithm is also shown to mitigate the limitation that the EASI algorithm can separate at most one Gaussian distributed source. Furthermore, the steady-state performance of the proposed algorithm is compared with that of EASI and the block-based second-order blind identification (SOBI) method

    Combining blockwise and multi-coefficient stepwise approches in a general framework for online audio source separation

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    This article considers the problem of online audio source separation. Various algorithms can be found in the literature, featuring either blockwise or stepwise approaches, and using either the spectral or spatial characteristics of the sound sources of a mixture. We offer an algorithm that can combine both stepwise and blockwise approaches, and that can use spectral and spatial information. We propose a method for pre-processing the data of each block and offer a way to deduce an Equivalent Rectangular Bandwith time-frequency representation out of a Short-Time Fourier Transform. The efficiency of our algorithm is then tested for various parameters and the effect of each of those parameters on the quality of separation and on the computation time is then discussed

    Model-Based Expectation-Maximization Source Separation and Localization

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    Convolutive Blind Source Separation Methods

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    In this chapter, we provide an overview of existing algorithms for blind source separation of convolutive audio mixtures. We provide a taxonomy, wherein many of the existing algorithms can be organized, and we present published results from those algorithms that have been applied to real-world audio separation tasks
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