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By
  
 
Humber ID-806970950

Submitted to:-
Dr.Ali Al-Rubaie
 Acknowledgement

We are very much thankful to    who is the lab instructor of our course Date
communication, for his value able guidance and advice without which we would not have been
able to complete the lab .this lab gave us an exposure to the industry as what we will be doing
there .his motivation contributed tremendously to the success of this lab.

We would also like to thank to  and our classmates and all professors for assisting us
in the collection of valuable information regarding the lab.Besides,we would also like to thank
the authority of Humber college for providing with such a good environment and facilities to
complete this lab.
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Analyze the frequency spectrum of sampled audio

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In signal processing,  !"# is the reduction of a continuous signal to a discrete signal. A

common example is the conversion of a sound wave (a continuous-time signal) to a sequence of
samples (a discrete-time signal).

A  ! refers to a value or set of values at a point in time and/or space.

A  ! is a subsystem or operation that extracts samples from a continuous signal. A

theoretical ideal sampler produces samples equivalent to the instantaneous value of the
continuous signal at the desired points.

For convenience, we will discuss signals which vary with time. However, the same results can be
applied to signals varying in space or in any other dimension and similar results are obtained in
two or more dimensions.

Let ( ) be a continuous signal which is to be sampled, and that sampling is performed by

measuring the value of the continuous signal every
seconds, which is called the sampling
interval. Thus, the sampled signal [] given bya

[] = (
), with  = 0, 1, 2, 3, ...

The sampling frequency or sampling rate  is defined as the number of samples obtained in one
second, or  = 1/
. The sampling rate is measured in hertz or in samples per second.

We can now ask: under what circumstances is it possible to reconstruct the original signal
completely and exactly (perfect reconstruction)?

A partial answer is provided by the NyquistShannon sampling theorem, which provides a

sufficient (but not always necessary) condition under which perfect reconstruction is possible.
The sampling theorem guarantees that bandlimited signals (i.e., signals which have a maximum
frequency) can be reconstructed perfectly from their sampled version, if the sampling rate is
more than twice the maximum frequency. Reconstruction in this case can be achieved using the
WhittakerShannon interpolation formula.

The frequency equal to one-half of the sampling rate is therefore a bound on the highest
frequency that can be unambiguously represented by the sampled signal. This frequency (half the
sampling rate) is called the Nyquist frequency of the sampling system. Frequencies above the
Nyquist frequency  can be observed in the sampled signal, but their frequency is ambiguous.
That is, a frequency component with frequency  cannot be distinguished from other components
with frequencies  +  and   for nonzero integers . This ambiguity is called aliasing. To
handle this problem as gracefully as possible, most analog signals are filtered with an anti-
aliasing filter (usually a low-pass filter with cutoff near the Nyquist frequency) before
conversion to the sampled discrete representation.

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As signals travel across a wire, certain factors will add noise to the signal.these factors can
include air conditioning units,magnetic fields.thus it require separating or filtering noise from
analog signals

"!$#"$!$#"
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a

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It contains 3 main blocks

Transmitter borad

Timing board

Receiver board


   

   

Transmitter borad also have 3 parts

"!$$a limit the analog frequency to about 3.4 khz .

 !"'(($: generates the samples of the analog input signals the sampling
pulse is connected to sample pulse input .the number of samples will depend upon the
frequency of the sampling pulses.

"#$#$(")$a this is a 4 bit counter type analog to digital converter.

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It is used to provide the sampling pulse and the analog signal

 + !!,a-the power suplly of 12 v is given by the generator.the signal

generator available in the lab was used to fedd this power.the pictorial view of the signal
generator is shown here under

 
*
(!athese leadsare used to feed the required produced signal on the screen
.these leads carry the signal from the board to the oscilloscope.

+!!,a-theseleads are used to feed the 12v power supply leads to the
equipment

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onnect 12 v D to the sample and hold board

Set the pulse generator to provide a pulse rate of 10 khz a pulse width of 10 microseconds
and 0 to +4 amplitude
alculate $,(,(-$ !m.!!
!.-&/.&//-&/0
onnect the scope to sampled out pin on the sample and hold board
Set the audio generator to 2 vpp at 500 hz and connect it to the filter in on the board .reset
the generator to 2 vpp when connected to the board and voltage level goes down
  !.-&/.1-/ !


 +"($'2"(,$&3'4 !-&/
 Set the analog frequency to 1.5 khz
Reduce the sampling rate to 6 khz