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DC - CH1 - Part1

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0% found this document useful (0 votes)
3 views35 pages

DC - CH1 - Part1

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© © All Rights Reserved
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ADC

Prepared by: Dr / Doaa Gamal


Lecturer at Faculty of Engineering, Suez Canal University
([email protected])
Pulse modulation
2

2
Pulse Code Modulation
3

 Most useful and widely used of PAM.

 Is a method of converting an analog signal


into a digital signal (A/D conversion).

 The conversion is performed by means of


sampling and quantizing.
PCM
4
SAMPLING
Sampling and reconstruction of analog signal
6

• Sampling rate (frequency) fs: number of samples per second.


• Sampling period Ts : time duration between two successive samples Ts =1/fs
Sampling rate
7
Sampling rate
8

 High sampling rate: higher quality of reconstructed signal,


higher Bw, higher storage.
 Very low sampling rate: bad quality of reconstructed
signal, low Bw, low storage. (cause aliasing effect:
distortion of the reconstructed signal)
 How to calculate the sufficient rate to avoid aliasing??
(Nyquist rate)
Sampling theorem
9

 The sampling theorem guarantees that an analog signal


can be in theory perfectly recovered as long as the
sampling rate, 𝑓𝑠 , is at least twice of the highest-frequency
component of the analog signal to be sampled, 𝑓𝑚𝑎𝑥 .
Otherwise, the signal cannot be reconstructed (aliasing
distortion).
𝑓𝑠 ≥ 𝑁𝑦𝑞𝑢𝑖𝑠𝑡 𝑟𝑎𝑡𝑒,
𝑁𝑦𝑞𝑢𝑖𝑠𝑡 𝑟𝑎𝑡𝑒 = 2𝑓𝑚𝑎𝑥 , 𝑓𝑠 = 1/𝑇𝑠
Sampling theorem
10
Sampling theory
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 Sampling in time domain leads to periodicity in frequency


domain and vice versa.
 The period in the frequency domain equals the sampling
frequency.
Sampling theorem
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• g(t) can be recovered from its samples by passing the sampled


signal 𝑔(𝑡) through a distortion-less LPF of bandwidth B Hz.
• The minimum sampling rate fs = 2B required to recover g(t) from
its samples 𝑔(𝑡) is called the Nyquist rate for g(t), and the
corresponding maximum sampling interval Ts = 1/2B is called
the Nyquist interval for the signal g(t).
Aliasing effect
13
Signal reconstruction
14

 Theoretically: Pass the sampled signal through an ideal


low-pass filter.
Signal reconstruction
15
Signal reconstruction problems
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1- sharp filter is unrealizable because


it has infinite time delay in the
response
Solutions:
- 𝑓𝑠 > 2𝐵 (gab between successive
cycles)
- Use a realizable LPF with gradual
cutoff characteristics

2- analog signal may not be bandlimitted


- Use anti-aliasing filter of the analog
signal before sampling to convert it to
band-limited signal
Signal reconstruction (practical)
17

Zero-order hold filter

The output is a stair-case


approximation of g(t)
PCM
18
QUANTIZATION
Uniform quantization
20

 m(t) is limited to the range (-mp, mp )


 The amplitude range is divided into ‘L’ uniformly spaced
levels each of width Δv = 2mp/L.
 A sampled point is approximated by the midpoint of the
interval in which it lies.
 The quantized samples are coded using a number of bits
equal (log 2 𝐿) and then transmitted as binary pulses.
Uniform quantization (example)
21

Example: if mpeak=1 V , number of output bits =8

𝐿 = 28 𝑙𝑒𝑣𝑒𝑙𝑠
2𝑚𝑝
∆𝑣 = = 7.81 𝑚𝑉
𝐿

- This implies that we can perfectly represent


analog voltages without error if they are
multiples of 7.81mV
- Otherwise, the intermediate values are
approximated to the nearest level, causing
quantization noise
Uniform quantization noise
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Example
23
24

24
25

Quantization Noise
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 The use of quantization introduces an error


between the I/P signal ‘m’ and the O/P
signal 𝑚 𝑡
q = m(t) –𝑚 𝑡 quantization error
 If ‘m’ is in the range ( - mp, mp )
Then the Step size of the quantizer :
Δv = 2 mp / L
26

Quantization Noise of uniform quantizer


26

The quantization noise is


𝑚 2
2 𝑝
2
𝑁𝑞 = 𝑞 = ∆𝑣
12 = 3𝐿2

So/No, the SNR of the quantizer,


Maximum information rate through a finite BW
27

 from the sampling theorem: a maximum of 2B independent


pieces of information per second can be transmitted, error free,
over a noiseless channel of bandwidth B Hz.
 Any sequence of independent data at the rate of 2B Hz can
come from uniform samples of a lowpass signal with
bandwidth B.
 a unit bandwidth (1 Hz) can transmit a maximum of two
pieces of information per second
28

Transmission bandwidth of a PCM wave


28

 Each encoded message sample is represented by a n-digit code


word.
 Each quantized sample is, thus, encoded into n bits. Because a signal
m(t) bandlimited to B Hz requires a minimum of 2B samples per
second, we require a total of 2nB bit/s, that is, 2nB pieces of
information per second. Because a unit bandwidth (1 Hz) can
transmit a maximum of two pieces of information per second, we
require a minimum channel of bandwidth BT Hz, given by
Nyquist theorem
29

Transmission bandwidth of a PCM wave


29
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Transmission bandwidth of a PCM wave


30
example
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Solution
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Non-Uniform quantization
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 Low amplitudes happens more frequently than larger


ones
 Since the mean square error is proportional to the step
size (Δv2 / 12), the noise power remains the same
regardless of the signal strength. we need to decrease
the step size for lower values than larger values.
 Nonuniform quantizers would improve SNR for low
signal amplitude and reduce SNR for large signal
amplitude. Such effect would lead to more balanced
overall quantization SNR that is less sensitive to input
signal strength than in case of uniform quantizer.
34

Non-Uniform quantization
34

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