The representation of

data in binary form.

The representation of
continuous signals as a
series of discrete
values.
The discrete
representation of sound
waves.
This is achieved by
converting the
sound energy by
means of a

transdu
cer (such as a
microphone) into
analogous electrical
variations of
current.

An

analog
to
digital
conver
ter
(ADC) is
then used to
periodically take
a series of
"snapshots" of the
voltage level of
the the analogous
signal.

This results in a
data stream of
numbers which
can be processed
and stored in
computer memory
or some other
storage media and
retrieved later to
reconstruct the
waveform using a

digital
to
analog
conver
ter

(DAC).

The rate at which the

"snapshots" of the signal
level are taken is known
as the sampling rate.
Sampling rate determines the bandwidth of
the resulting digital representation with a rate of twice the highest
frequency desired being required.

The Nyquist theorem states that the

sampling rate must be greater than
2X the highest frequency in the input
signal.

The Nyquist frequency is 1/2 the

sampling rate.
The continuous representation of
sound waves.

Digital filters are a very important part of DSP. In fact, their extraordinary performance is one of
the key reasons that DSP has become so popular. As mentioned in the introduction, filters have
two uses: signal separation and signal restoration. Signal separation is needed when a signal has
been contaminated with interference, noise, or other signals. For example, imagine a device for
measuring the electrical activity of a baby's heart (EKG) while still in the womb. The raw signal
will likely be corrupted by the breathing and heartbeat of the mother. A filter might be used to
separate these signals so that they can be individually analyzed.
Signal restoration is used when a signal has been distorted in some way. For example, an audio
recording made with poor equipment may be filtered to better represent the sound as it actually
occurred. Another example is the deblurring of an image acquired with an improperly focused
lens, or a shaky camera.

Linear filter:

Additivity: f(x + y) = f(x) + f(y).

Homogeneity of degree 1: f(x) = f(x) for all .

Casual filter:
a causal filter is a linear and time-invariant causal system
The word causal indicates that the filter output depends only on past and present
inputs

Time invariance means that whether we apply an input to the system now or T
seconds from now, the output will be identical except for a time delay of T seconds.
That is, if the output due to input

is

, then the output due to input

is
. Hence, the system is time invariant because the output does not
depend on the particular time the input is applied.

FIR filter:
In signal processing, a finite impulse response (FIR) filter is a filter whose
impulse response (or response to any finite length input) is of finite duration,
because it settles to zero in finite time

For a causal discrete-time FIR filter of order N, each value of the output sequence is a weighted
sum of the most recent input values:

where:

is the input signal,

is the output signal,

is the filter order; an

th-order filter has

terms on the right-hand side

is the value of the impulse response at the i'th instant for

of an th-order
FIR filter. If the filter is a direct form FIR filter then is also a coefficient of the filter .

This computation is also known as discrete convolution.

IIR filter

Systems with this property are known as IIR systems or IIR filters, and are
distinguished by having an impulse response which does not become exactly zero
past a certain point, but continues indefinitely.

Advantages and disadvantages

The main advantage digital IIR filters have over FIR filters is their efficiency in implementation,
in order to meet a specification in terms of passband, stopband, ripple, and/or roll-off. Such a set
of specifications can be accomplished with a lower order (Q in the above formulae) IIR filter
than would be required for an FIR filter meeting the same requirements. If implemented in a
signal processor, this implies a correspondingly fewer number of calculations per time step; the
computational savings is often of a rather large factor.
On the other hand, FIR filters can be easier to design, for instance, to match a particular
frequency response requirement. This is particularly true when the requirement is not one of the
usual cases (high-pass, low-pass, notch, etc.) which have been studied and optimized for analog
filters. Also FIR filters can be easily made to be linear phase (constant group delay vs frequency)
a property that is not easily met using IIR filters and then only as an approximation (for
instance with the Bessel filter). Another issue regarding digital IIR filters is the potential for limit
cycle behavior when idle, due to the feedback system in conjunction with quantization.