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Low bit-rate speech coding based on multicomponent AFM signal model

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Abstract

In this paper, we propose a novel multicomponent amplitude and frequency modulated (AFM) signal model for parametric representation of speech phonemes. An efficient technique is developed for parameter estimation of the proposed model. The Fourier–Bessel series expansion is used to separate a multicomponent speech signal into a set of individual components. The discrete energy separation algorithm is used to extract the amplitude envelope (AE) and the instantaneous frequency (IF) of each component of the speech signal. Then, the parameter estimation of the proposed AFM signal model is carried out by analysing the AE and IF parts of the signal component. The developed model is found to be suitable for representation of an entire speech phoneme (voiced or unvoiced) irrespective of its time duration, and the model is shown to be applicable for low bit-rate speech coding. The symmetric Itakura–Saito and the root-mean-square log-spectral distance measures are used for comparison of the original and reconstructed speech signals.

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Authors and Affiliations

  1. Department of Electrical Engineering, Indian Institute of Technology Kanpur, Kanpur, Uttar Pradesh, 208016, India

    Mohan Bansal & Pradip Sircar

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  1. Mohan Bansal
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  2. Pradip Sircar
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Correspondence to Mohan Bansal.

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Bansal, M., Sircar, P. Low bit-rate speech coding based on multicomponent AFM signal model. Int J Speech Technol 21, 783–795 (2018). https://doi.org/10.1007/s10772-018-9542-5

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