Sampling of Speech Signals

Review of the Statistical Model for Speech

SAMPLING OF SPEECH SIGNALS
Speech is a continuous function of continuous
time variable.
It is sampled periodically in time to produce a
sequence of samples X(nT).
It is necessary to quantize these sample
values to a finite set of values in order to obtain
digital representation.
Since we are concerned with digital
representation of speech signals we need to
consider spectral properties of speech.
According to steady state models for the
production of vowel and fricative sounds,
speech signals are not inherently band
limited.
Spectrum tend to fall off rapidly at high
frequencies
Thus to accurately represent all speech
sounds would require a sampling rate greater
than 20KHz
REVIEW OF THE STATISTICAL MODEL FOR
SPEECH
Speech waveform can be represented by an
ergodic random process.
Assume that the signal x(t) is a sample
function of continuous time random process
then the sequence of samples obtained by
sampling can be thought of as a sample
sequence of discrete time random process.

The power spectrum of sampled signal is an
aliased version of the power spectrum of the
original signal.
The averages such as mean and variance
are the same for samples as for the original
signal.



PROBABILITY DENSITY ESTIMATION
It is estimated by determining a histogram of
amplitudes for a large number of samples.
A good approximation to measure probability
density is gamma distribution given as

Similar approximation is the laplacian density



The auto correlation function and power spectrum of
speech signals can be estimated by standard time
series analysis methods
An estimate of the autocorrelation function can be
obtained by estimating the time average
autocorrelation function from a long segment of a
signal.
The power spectrum can be estimated in a variety of
ways
1. By measuring the average output of the set of
bandpass filters.
2. Estimation of long term average power
spectrum
3. Computing the power transfer function of a
recursive digital filter