Analog To Digital Conversion Digital To Analog Conversion: Tanauan City College
Analog To Digital Conversion Digital To Analog Conversion: Tanauan City College
Province of Batangas
CITY OF TANAUAN
Instead of a pulse train, PCM produces a series of numbers
or digits, and hence this process is called as digital. Each
one of these digits, though in binary code, represent the
approximate amplitude of the signal sample at that instant.
In Pulse Code Modulation, the message signal is
represented by a sequence of coded pulses. This message
signal is achieved by representing the signal in discrete
form in both time and amplitude.
Components for PCM method
Basic Elements of PCM
The transmitter section of a Pulse Code Modulator circuit consists
of Sampling, Quantizing and Encoding, which are performed in the
analog-to-digital converter section. The low pass filter prior to sampling
prevents aliasing of the message signal.
The basic operations in the receiver section are regeneration of
impaired signals, decoding, and reconstruction of the quantized
pulse train. Following is the block diagram of PCM which represents the
basic elements of both the transmitter and the receiver sections.
Low Pass Filter
This filter eliminates the high frequency components present in the input
analog signal which is greater than the highest frequency of the message
signal, to avoid aliasing of the message signal.
Sampler
This is the technique which helps to collect the sample data at instantaneous
values of message signal, so as to reconstruct the original signal. The
sampling rate must be greater than twice the highest frequency
component W of the message signal, in accordance with the sampling
theorem.
Quantizer
Quantizing is a process of reducing the excessive bits and confining the
data. The sampled output when given to Quantizer, reduces the redundant
bits and compresses the value.
Encoder
Decoder
The decoder circuit decodes the pulse coded waveform to reproduce the original
signal. This circuit acts as the demodulator.
Reconstruction Filter
After the digital-to-analog conversion is done by the regenerative circuit and the
decoder, a low-pass filter is employed, called as the reconstruction filter to get
back the original signal.
Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it
and samples it, and then transmits it in an analog form. This whole process is
repeated in a reverse pattern to obtain the original signal.
Sampling is the process of converting analog signal into a
discrete signal or making an analog or continuous signal to occur
at a particular interval of time, this phenomena is known as
sampling.
SAMPLING THEOREM:-
Sampling theorem states that a band limited signal having no
frequency components higher than fm hertz can be sampled if
its sampling freq is equal to or greater than Nyquist rate.
Sampling Techniques
Their are basically three types of Sampling
techniques, namely:
1. Natural Sampling
2. Flat top Sampling
3. Ideal Sampling
The following figure indicates a continuous-
time signal x tt and a sampled signal xs tt.
When x tt is multiplied by a periodic impulse
train, the sampled signal xs tt is obtained.
Sampling Rate
To discretize the signals, the gap between the samples
should be fixed. That gap can be termed as a sampling
period Ts.
For an example of the Nyquist theorem, let us sample a simple sine wave at three sampling rates: f s =
fs = f (one-half the Nyquist rate). Figure 4.24 shows the sampling and the subsequent recovery of the
signal.
It can be seen that sampling at the Nyquist rate can create a good approximation of the original sine
wave (part a). Oversampling in part b can also create the same approximation, but it is redundant and
unnecessary. Sampling below the Nyquist rate (part c) does not produce a signal that looks like the
this signal?
Solution
The bandwidth of a low-pass signal is between 0 and f, where f is the maximum frequency in the
signal. Therefore, we can sample this signal at 2 times the highest frequency (200 kHz). The
We want to digitize the human voice. What is the bit rate, assuming 8 bits per sample?
Solution
The human voice normally contains frequencies from 0 to 4000 Hz. So the sampling rate and bit
4.31
Delta demodulation components
Digital Specifications
• The relationship between analog
signals and their digital equivalent is
important to the overall
performances of a system.
4.33
Factors that determine the
relationship between an analog and
its corresponding digital value
1. Accuracy
2. Resolution
4.34
Accuracy
4.35
Resolution
4.36
Quantization
• Is the process of converting the
sampled signal to a binary value
• Each voltage level will correspond to
a different binary number
v
1.0
¼ t
0 to 1 V- analog signal
0
4.38
v
111
1.0
110
101
¾
100
011 ½
010
001
¼ 0 to 1 V- analog signal t
The same signal with a resolution of 1 in 8 parts
000 (three bits)
0
4.39
v
11 1.0
¾
10
½
01
¼ 0 to 1 V- analog signal t
00 The same signal with a resolution of 1 in 4 parts
0 (two bits)
4.40
Pulse Code Modulation
The size of each division or magnitude of the
minimum step is called the STEP SIZE
Quantization Error (Quantization Noise)
- is used to describe that a digital value
corresponds to a distinct span of analog signals.
To reduce the quantization error, more divisions
(higher resolution)are used to represent the
analog value.
4.42
Dynamic Range
This is the ratio of the largest to smallest
It follows that
If this is expressed in decibels
From