AUTOCORREALTION

The autocorrelation function is defined as the measure of similarity or coherence between a signal and its time
delayed version. Therefore, the autocorrelation is the correlation of a signal with itself.

The autocorrelation function is often used as a means of detecting periodicity in signals

If the signal is ramdaom or periodic the appropriate definition is

The quantity ∅(�) =

�� ����� �� �ℎ�� ������ ��� ������������� ������� �� ������� ����� ��� periodic
signals 1

Short-Time Auto-Correlation Function
Short-Time Auto-Correlation Function

(0.008438 – 0.00275)*Fs = 91 samples

Pitch Period = 0.0057 sec, Pitch Freq = 175.8 Hz
Short-Time Average Magnitude Difference Function
Short-Time Average Magnitude Difference Function

0.008376-0.002488 = 0.0059 sec

0.0059sec*Fs = 94 samples
UNIT –III:

Linear Predictive Coding (LPC) Analysis: Basic principles of Linear Predictive Analysis, computing the
prediction coefficients: auto correlation, formulation of Linear prediction coding equations, Solution of Linear
prediction coding equations; The Levinson Durbin recursion, LPC spectrum, Applications of LPC Parameters:
Pitch Detection using LPC Parameters.

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Linear Time invarient Discreate time system characterized
by
In general, an FIR system is described by the diffrence equation

After applying Z transform

Direct-form realization of FIR System
 Linear Predictive Analysis
qLinear predictive analysis is one of the most powerful and widely used speech analysis
techniques.

qThe importance of this method lies both in its ability to provide accurate estimates of the
speech parameters and in its relative speed of computation.

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This method becomes the prominent technique for estimating the basic
paramanters ,e.g pitch, formats, spectr. vocal tract area functions, and
representing speech for low bit rate transmission or storage

The basic idea behind linear predictive analysis is that a speech sample can
be approximated as a linear combination of past speech samples.

Minimizing the sum of squared differecnes between the actual speech

samples an sthe linearly predicted ones, a unique set of predictors coeffients
can be dtermined.

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 Basic Priciple of Linear Predictive Analysis
The basic discrete -time model for speech production model is shown in Fig.

In this , The Composite spectrum spectrum effects of radiation Vocal Tract

and Glottal Excitation are represented by a time varying digital filter whoxse
steady-state system function is od the form

This system is excited by an impulse train for voice speech or a random noise sequence for unvoiced speech. , Thus the
parameters of this model are voice/unvoice classfifcation ,pitch period for voice speech , The parameters to be
calculated are G and {aK} of a digital filter. these are also vary with time

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The pitch period and voice/unvoiced classiifcation can be estimated using Linear predictive method. If the order p
is high enough, the all pole model provides a good representation for almost all the sounds of speech. The major
advantage of this model is that the gain papramater G, and the filter coefficients {ak} can be estmimated very
strightforward and computat analysis.

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 Frequency domain repesentation of Speech production model

For the speech production model , the speech sampels S(n) are related to the excitation u(n) by the simple time
domain equation derived from H(z)
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Between pitch pulses Gu(n) is zero.therefore s(n) can be predicted from a
linearly weigted summatation of past samples. Gu(n) is included then we can
predict s(n) only aprroximatly.

A linear predictor with prediction coeffients , ak is defined as system

The prediction error,e(n) is defiend as

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The prediction error sequence is the output of a system whose tranfer function is

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 From the equation and

The speech signal obeys only when ak and αk then e(n)=Gu(n). The prediction error filter A(Z) will be inverse filter
for the system H(z) . Thus prediction error filter A(z), will be an inverse filter for the system H(z).

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q The time varying nature of the speech sinal the predictor coeffients must be edstimated
from short segements of the speech signal.

q The basic aThe basic linear prediction analysis is to determine a set of predictor coeffients {αk}
directly from the speech signal in a sucha manner as to obtain a good estimate of the sprectral
propeprties of the speech signal.

q pproach that will minimize the the mean squared prediction error over a segmemtn of the
speech waveform . The resulted parameters are assumed to be the paramters of the
system function H(z).

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The short time average prediction error is defiend as

Where Sn(m) is a segment of speech that has been selcted in the vicity of sample n

The range of summation is templorily left unspeciifed but we wish to develop a short time analysis technique,
the sum will always be over a finite interval.
��
To determine the values of ∝� that minimize �� by setting � =0, i=1,2,......p
�∝�

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���
=0, i=1,2,......p
�∝�

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This set of equations in p unkowns can be solved in an efficent manenr for the unknown predictor
coeffients {�� } that mimimize the avaerage squared prediction erro for the segment �� (m) , the
mimimum mean squared prediction error can be shown to be

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 The autocorrelation method
�� (�) �� ����������� ���� ������� �ℎ� �������� 0 ≤ � ≤ � − 1
This can be conviently expressed as �� (�) =�(� + �)�(�)
The limits on the expression for ∅� (�, �)

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The PXP MATRIX OF AUTO CORRELATION VALUE IS tOEPLITZ MATRIX ,i.e Symettric and all the
elements along a given diagonal are equal.

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Computation of the Gain for the model
It is reasonable to expect that the gain G, could be determined by matching the energy in the signal with the energy
of the linearly predicted samples. This indeed is true when approproiate assumptions are made about the excitations
sinal to the LPC system.
It is possible to relate the gain constant G to the excitation singal and the error in prediction by referring

The prediction error is represented by

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The input singal is propotional to thw error signal with the cosntant of propotionality being the gain
constant G.
It is generally not possible to solve for G in a realiable way direlty from the eeror signal itself

Insted the more reasonable assumption is made that the energy in the error signal is equal to the enery in
the excitation input

At this point we must make some assumptions about u(n) so as to be able to relate G to the known
quntities �� .

For voiced it is reasonable to assume u(n)=impulse function (excitation is at n=0)

For unvoiced is reasonable to assume cosntant function i.e u(n) is a zero mean, unity varience white
noise signal
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The resulting output for this particular input h(n)

The auto corrwlation function h(n)

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Short-Time Auto-Correlation Function

(0.008438 – 0.00275)*Fs = 91 samples

Pitch Period = 0.0057 sec, Pitch Freq = 175.8 Hz
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 Durbins recursive solution for the auto correlation
Equations

All the elements on a given diagonal in the matrix are equal.

The auro correlation method , the matrix equation for solving the predictior equations is of the form

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By explaiting the Toeplitz nature of the matrix of coeffients, seveal effiecient
recursive procedures have been devised for solving this system of equations .
Most Popular procedure which can be stated as follows.

The above equations are soved recursively for i=1,2....p and final solution is given as

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To issustrate the procedure consider an example of obtaning the predictor coeffients for a predictor of order 2

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LPC spectrum

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comparison between short-time Fourier analysis and linear
predictive spectrum analysis for segments of voiced and
unvoiced speech.

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