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Analog to Digital Conversion

The document discusses the process of converting analog signals to digital format through Analog to Digital Conversion (ADC), specifically using Pulse Code Modulation (PCM). It outlines the key steps involved in PCM, including sampling, quantizing, and encoding, and emphasizes the advantages of digital systems over analog ones, such as improved reliability and the ability to regenerate clean signals. Additionally, it explains the importance of sampling rates and the Nyquist theorem in ensuring accurate signal reconstruction.

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Sh Rahat
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2 views

Analog to Digital Conversion

The document discusses the process of converting analog signals to digital format through Analog to Digital Conversion (ADC), specifically using Pulse Code Modulation (PCM). It outlines the key steps involved in PCM, including sampling, quantizing, and encoding, and emphasizes the advantages of digital systems over analog ones, such as improved reliability and the ability to regenerate clean signals. Additionally, it explains the importance of sampling rates and the Nyquist theorem in ensuring accurate signal reconstruction.

Uploaded by

Sh Rahat
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
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Analog to Digital

Conversion
By:
Md. Ibrahim Khalil
M.Sc in ICT, BUET (Ongoing)
B.Sc in ETE, CUET
What do you think might be an example
of an analog system in action?
“I still have no idea what that stuff is!”

There’s probably a reason for that. We don’t really


use many systems that are completely analog
anymore; digital communications are more widely
used.
Interesting fact and important to the class:

□ Digital signals are not representative of


signals that occur in nature.

□ Natural signals are analog, and must be


converted into digital format to be used
in a digital system
Great! So we’re using a new method to get
the same information. Is this a big deal?

❑ It is, because using digital systems


offers a number of advantages over
using analog systems.
Integrating Aspects of Multimedia
Audio/Video
Image/Video
Presentation
Capture Audio/Video Playback
Perception/
Playback
Image/Video Information
Representation

Transmission Transmission

Compression
Audio Processing
Capture

Audio Information Media


A/V
Representation Server
Playback
Storage
CS 414 - Spring 2012
Why digital?

In analog system, a message signal suffers the channel


noise and the signal distortion, which are cumulative.

Amplification is of little help because it enhances the


signal and the noise in the same proportion.

The analog signal can not cleaned periodically, and


thus the transmission is not reliable.
Why digital?

In Digital system, the new, clean signals can be


completely regenerated at repeater stations
because all the information is contained in the
code.

The Digital signal can then be transmitted over


long distance with great reliability.
So, How Analog to Digital?

There have a technique,

Pulse Code Modulation


(PCM)
Pulse Code Modulation (PCM) is a technique to
convert analog data to digital signal.

The PCM is generated by carrying out 3 basic signal


operations:

1. Sampling
2. Quantizing
3. Encoding.

12
Components of PCM encoder
Sampling

Sampling is the processes of converting


continuous-time analog signal, xa(t), into a
discrete-times by taking the “samples” at
discrete-time intervals

– Sampling analog signals makes them discrete in


time but still continuous valued.
– If done properly (Nyquist theorem is satisfied),
sampling does not introduce distortion.
Sampling
Sampled values:
– The value of the function at the sampling points

Sampling interval:
– The time that separates sampling points (interval b/w
samples), Ts
– If the signal is slowly varying, then fewer samples per
second will be required than if the wave is rapidly
varying
– So, the optimum sampling rate depends on the
maximum frequency component present in the signal.
Sampling
- Sampling Rate (or sampling frequency fs):
- Sampling rate at which the signal is sampled, expressed
as the number of samples per second reciprocal of the
sampling interval), 1/Ts = fs
Nyquist Sampling Theorem (or Nyquist Criterion):

- The sampling is performed at a proper rate, no


info is lost about the original signal and it can be
properly reconstructed later on
Statement:
“If a signal is sampled at a rate at least, but not
exactly equal to twice the max frequency
component
of the waveform, then the waveform can be exactly
reconstructed from the samples without any
distortion”
f >2f
Sampling…

- Sampling Theorem for Bandpass Signal - If an analog


information signal containing no frequency outside the
specified bandwidth W Hz, it may be reconstructed
from its samples at a sequence of points placed 1/(2W)
seconds apart with zero-mean squared error.

The minimum sampling


rate of (2W) samples per
second, for an analog
signal bandwidth of W Hz,
is called the Nyquist rate.
Sampling rate

Sampling rate is defined as the number of samples taken


per second from a continuous signal for a finite set of
values. We can also define it as a sampling frequency,
which is the reciprocal of the sampling time.

FS = 1/ TS
Where, FS is the sampling frequency
TS is the sampling time

sampling rate is an essential period for the sampler to


perform sampling process. It helps in the successful
recovery of the digital signal at the receiving end.
Nyquist Interval

Nyquist interval is the reciprocal of the Nyquist


rate. It is given by:
TS = 1/2W

Where, TS is the Nyquist Interval


W is the highest frequency
Methods of sampling

The methods of sampling are classified as follows:

□ Ideal Sampling
□ Natural Sampling
□ Flat-top sampling
Ideal Sampling

The sampling process multiplies the input signal and the


carrier signal, which is present in the form of train of
pulses.
Natural Sampling

Natural Sampling is considered an efficient multiplexing


method in Pulse Amplitude Modulation. Here, the analog
signal is multiplied by the uniformly spaced rectangular
pulses.

xs(t) = x(t)xp(t)
Flat-top sampling

The design and reconstruction of flat-top sampling is easy


than the natural sampling process. The pulses in the
flat-top sampling method are in the flat shape at the top
and are held at a constant height. It means that the
samples are flat and have constant amplitude.
1. Sampling

Fig. 2: name
2. Quantized PAM Signal
• Quantization: is a method of assigning integer values in a specific
range to sampled instances.

Fig. 3: name
4.27

Quantization
Sampling results in a series of pulses of varying
amplitude values ranging between two limits: a min and
a max.

The amplitude values are infinite between the two


limits.

We need to map the infinite amplitude values onto a


finite set of known values.

This is achieved by dividing the distance between min


and max into L zones, each of height Δ.
Δ = (max - min)/L
4.28

Quantization Levels

□ The midpoint of each zone is assigned a


value from 0 to L-1 (resulting in L values)

□ Each sample falling in a zone is then


approximated to the value of the midpoint.
4.29

Quantization Zones

Assume we have a voltage signal with amplitutes Vmin=-20V and

Vmax=+20V.

We want to use L=8 quantization levels.

Zone width Δ = (20 - -20)/8 = 5

The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to 0, 0 to +5, +5

to +10, +10 to +15, +15 to +20

The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5, 17.5
3. Binary Encoding

- Each quantized samples is translated into equivalent binary codes .

Table 1: name
4.31
Figure 4.26 Quantization and encoding of a sampled signal
32

Example
amplitude
x(t)
111 3.1867

110 2.2762 Quant. levels


101 1.3657

100 0.4552

011 -0.4552 boundaries

010 -1.3657

001 -2.2762 x(nTs): sampled values


xq(nTs): quantized values
000 -3.1867
Ts: sampling time
PCM t
codeword 110 110 111 110 100 010 011 100 100 011 PCM sequence
33
Quantization Effect

• Sampling and Quantization Effects


▫ Quantization (Granularity) Noise: Results when
quantization levels are not finely spaced apart
enough to accurately approximate input signal
resulting in truncation or rounding error.
▫ Quantizer Saturation or Overload Noise: Results
when input signal is larger in magnitude than
highest quantization level resulting in clipping of
the signal.
▫ Timing Jitter: Error caused by a shift in the
sampler position. Can be isolated with stable clock
reference.
4.34
Assigning Codes to Zones

Each zone is then assigned a binary code.


The number of bits required to encode the zones, or
the number of bits per sample as it is commonly
referred to, is obtained as follows:
nb = log2 L
Given our example, nb = 3
The 8 zone (or level) codes are therefore: 000, 001,
010, 011, 100, 101, 110, and 111
Assigning codes to zones:
000 will refer to zone -20 to -15
001 to zone -15 to -10, etc.
Rb = bit rate = No. of bit used to encode a sample * sampling Rate
* Rb = nfS
* L = 2n fS = Sampling frequency
* fNQ = 2 fm
fm = message signal’s frequency
fNQ = Nyquist frequency
• fS ≥ fNQ
L = No. of Quantization level
n = No. of bits in the sample’s code

* Standard sampling rate for telephone system = 8KHz


* We can transmit up to 2 bit/s with 1Hz Bandwidth.
Example
Example
Example
Thank you all.

Questions?

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