Fourier transform could be devided into four basic forms base don tyupe of signal obtined

1.
2.
3.
4.

Aperiodic-contrinuous
Periodic-continuous
Aperiodic-Discrete
Periodic-Discrete

Foureier Transform
Fourier Series
Discrete time fourier Trasnsfomr
Discrete Foureier trasnsform

There is no Foureier trasform that uses finite length signal. The way around is imagining all the other
values of the signal are zeros or to replicate the same set of data infinite times.
Padding with zeros makes the signal descrete aperiodic and we need Descrete time Fourier
Transform, instead if we imagine that signal is replicated then we can use DFT.
Inverse DTFT need infinite sinusoids to reconstruct eh signal back, which is not possible in
computers practically hence all computer alfotrithms use DFT only considering the obtained
signal is descrte peroiic signal.
Each fourier Transform can be sub divided into real and complex.
Transforms are direct extentions of functions allowing both the input and output to have multiple
values. Size of RHS need not match with LHS, signals could be continuous, descrete, real complex,
mixed any thing. In short trasnsomr is a fixed procedure that changes one chunk of data into
another chunk of data.
Here DFT changes N point input signal into two( N/2 +1) point output signals The input signal
contains the signal being decomposed, while the two ouput singals contain the scaled amplitudes
of the component sin and cosine waves. The input signal is said to be in time domain as we
obtain the samples on time basis, and out put is said to be in frequency domain as we obtain the
amplitudes of sin and cosine waves of different frequencies.
The frequency domain contains exactly the same informaton as the time domain just in different
form.
Number of samples in the time domain is useally represented by variable N. N will beusually in
powers of 2. i.e, 128,256,512 etc. as digital data storeage uses binary addressing, making power
of two a natural signal length and FFT algorithm usually operated with N that is a power of 2.
Lower case letters for time domain x[],y[]. And uper case for frequency domain X[],Y[].
X[] is said to have two parts ReX[] and ImX[], which are actually amplitudes of cosine and sin waves
respectively.
Below figure shows How a 16 point time domain data appears when it is decomposed to 9 + 9 cos
and sin waves of different amplitudes.

Note 1: In the above picture why I am trying to express with only N/2 = 8 different frequency
components? Whay not more. In the above example why I started at 1 cycle/16 samples and

stoped at 8 cycles/16 samples. Cant I take more components of higher frequencies ? If so what
happens? Hint: Sampling Theorum.
Note2: adding all these amplitudes should give us the original signal back again.
Frequency domains independent variable:
Above decomposition will be represented in frequency domain. Below figure shows how a 127 point
time domain data is represented in frequency domain. 64 amplitudes of cos components and 64
amplitudes of sine components.

The horizontal axis of the frequency domain can be referred in four different ways, all of which are
common in DSP.
First method horizontal axis is labeled from 0 to 64, corresponding to the 0 to N/2 samples in the
arrays.
Second method is the horizontal axis is labeled as fraction of the sampling rate. This means that the values
along the horizontal axis always run between 0 and 0.5, since discrete data can only contain frequencies
between DC and one-half the sampling rate. These values of 0 to 0.5 are known as normalized frequencies f =
k/N. where k is sample index running from 0 to N/2. The third style is similar to second except the horizontal
axis is multiplied by 2. The index used with this labeling is , a lower case Greek omega. Note here omega

represents discrete time frequency. The last method is easy to write. Consider how one of the cosine
components could be written in four different ways c[n] = cos(2kn/N), c[n] = cos(2fn) & c[n] = cos(wn).
Fourth way is to multiply sampling rate to normalized frequency to get analog frequencies. For instance, if the
system being examined has a sampling rate of 10 kHz graphs of the frequency domain would run from 0 to 5
kHz.
Correlation:

In order for this correlation algorithm to work, the basis function must have an interesting property:
each of them must be completely uncorrelated with all of the others. This means that if you multiply any
two of the basis function the sum of resulting points will be equal to zero. Basis functions that have this
property are called Orthogonal.
Frequency response of Systems:
Systems are analyzed in the time domain by using convolution. A similar analysis can be done in the frequency
domain. Using the Fourier transform, every input signal can be represented as a group of cosine waves, each
with a specified amplitude and phase shift. Likewise, the DFT can be used to represent every output signal in a

similar form. This means that any linear system can be completely described by how it changes the amplitude
and phase of cosine waves passing through it. This information is called the system's frequency response.
Since both the impulse response and the frequency response contain complete information about the system,
there must be a oneto- one correspondence between the two. Given one, you can calculate the other. The
relationship between the impulse response and the frequency response is one of the foundations of signal
processing: A system's frequency response is the Fourier Transform of its impulse response. Figure 9-6
Illustrates these relationships.