Advances in Non-Linear Modeling for Speech Processing by Raghunath S. Holambe

By Raghunath S. Holambe

Advances in Non-Linear Modeling for Speech Processing comprises complicated subject matters in non-linear estimation and modeling suggestions in addition to their purposes to speaker attractiveness.

Non-linear aeroacoustic modeling process is used to estimate the real fine-structure speech occasions, which aren't printed by way of the fast time Fourier remodel (STFT). This aeroacostic modeling technique offers the impetus for the excessive answer Teager power operator (TEO). This operator is characterised by way of a time solution that may song swift sign strength adjustments inside a glottal cycle.

The cepstral gains like linear prediction cepstral coefficients (LPCC) and mel frequency cepstral coefficients (MFCC) are computed from the importance spectrum of the speech body and the section spectra is overlooked. to beat the matter of neglecting the section spectra, the speech construction procedure will be represented as an amplitude modulation-frequency modulation (AM-FM) version. To demodulate the speech sign, to estimation the amplitude envelope and on the spot frequency parts, the power separation set of rules (ESA) and the Hilbert remodel demodulation (HTD) set of rules are mentioned.

Different positive aspects derived utilizing above non-linear modeling thoughts are used to advance a speaker identity method. ultimately, it really is proven that, the fusion of speech construction and speech belief mechanisms may end up in a powerful characteristic set.

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This model can be used in two modes, • For prediction: For voiced speech, e[n] can be thought of as a train of unit samples. , every pitch period, from Eq. , N ak x[n − k], when e[n] = 0. 3) k =1 The system function associated with the N th order predictor is a finite length impulse response (FIR) filter of length N given as N ak z −k . 4) k =1 • For signal synthesis or modeling: Where the prediction residuals are important. 5) k =1 and the associated prediction error filter is defined as N A(z) = 1 − ak z −k k =1 = 1 − P(z).

Under this coding scheme, the parameters of the LPC model need to be estimated for each separate block of speech. Such estimated LPC parameters (together with other residual information) are used to represent the original speech samples on a frame-by-frame basis. These estimated parameters have also been used as speech feature vectors for speech recognition. A more efficient way to characterize the time dependency of the parameter vector θ in the LPC model is to provide a parametric form for the parameter evolution [19].

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