Programmer Guide/Command Reference/EVAL/lpc: Difference between revisions
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:;<var>type</var>:output selector; 0 <= ''type'' <= 4 (default=0) | :;<var>type</var>:output selector; 0 <= ''type'' <= 4 (default=0) | ||
:;<var>lfft</var>:the length of the fft to be used for the computation of the transfer function (amplitude spectrum) of the inverse filter; ''m+1'' < ''lfft'' | :;<var>lfft</var>:the length of the fft to be used for the computation of the transfer function (amplitude spectrum) of the inverse filter; ''m+1'' < ''lfft'' | ||
:note: if necessary, the value of ''lfft'' is automatically corrected to the next possible value | ::note: if necessary, the value of ''lfft'' is automatically corrected to the next possible value | ||
;Description: | ;Description: | ||
:# apply differentiation to signal ''x'' | :# apply differentiation to signal ''x'' |
Revision as of 14:20, 12 April 2011
Compute the linear prediction coefficients using the autocorrelation method. This function implements an all-pole filter approximation of the acoustic tube modell of the vocal tract. The implementation is based on the book "Linear Prediction of Speech" (J.D. Markel & A.H. Gray, Springer 1976).
- Usage
lpc(x, m, p {, type {, lfft}})
- x
- the signal vector; this should be a speech signal without windowing function (because the hamming-window is applied to x by this the function)
- m
- number of coefficients
- rule of thumb: m ~ samplingrate / 1000 * 1.25
- p
- differentiation factor; 0 <= p <= 1 (default=0)
- type
- output selector; 0 <= type <= 4 (default=0)
- lfft
- the length of the fft to be used for the computation of the transfer function (amplitude spectrum) of the inverse filter; m+1 < lfft
- note: if necessary, the value of lfft is automatically corrected to the next possible value
- Description
-
- apply differentiation to signal x
- apply hamming window to signal x
- use the autocorelation method to compute the inverse filter coefficients ai, the reflection coefficients rc and the error (or residual) energy alpha
- convert coefficients to the selected result
- Result
- A vector y containing the result of the function.
type y nrow(y) description 0 amplitude spectrum of the inverse filter lfft/2+1 this function can be used in speech analysis to compute the transfer function of the vocal tract (e.g. for formant extraction) 1 y[0]=alpha
y[1..M+1]=ai[0..m]
m+2 the error energy (alpha) and the m+1 inverse filter coefficients ai 2 y[0]=alpha
y[1..M]=rc[0..m-1]
m+1 the error energy (alpha) and the m reflection coefficients rc 3 y[0]=alpha
y[1..M+1]=ar[0..m]
m+2 the error energy (alpha) and the m area coefficients ar
(ar[i] ~ area of section i)4 y[0]=alpha
y[1..M+1]=lar[0..m]
m+2 the error energy (alpha) and the m log. area coefficients lar
(lar[i] ~ diameter of section i)