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Lec 15

This lecture introduces Frequency Shift Keying (FSK), a digital modulation technique based on orthonormal pulses that differ in frequency. The transmitted signal for bits 0 and 1 is derived from these pulses, and the lecture discusses the received signal and the need for a matched filter to optimize signal reception. The derivation of the receive filter for optimal Signal-to-Noise Ratio (SNR) is also presented, with a promise to explore SNR and bit error rate in the next module.

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0% found this document useful (0 votes)
3 views15 pages

Lec 15

This lecture introduces Frequency Shift Keying (FSK), a digital modulation technique based on orthonormal pulses that differ in frequency. The transmitted signal for bits 0 and 1 is derived from these pulses, and the lecture discusses the received signal and the need for a matched filter to optimize signal reception. The derivation of the receive filter for optimal Signal-to-Noise Ratio (SNR) is also presented, with a promise to explore SNR and bit error rate in the next module.

Uploaded by

debjani goswami
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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Principles of Communication Systems – Part II

Prof. Aditya K. Jagannatham


Department of Electrical Engineering
Indian Institute of Technology, Kanpur

Lecture - 15
Introduction to Frequency Shift Keying (FSK)

Hello, welcome to another module in this massive open online course. In this module, let
us look at another digital modulation technique that is frequency shift key.

(Refer Slide Time: 00:25)

So, we will be looking at a new digital modulation scheme which is termed as frequency
shift key. This is a digital modulation scheme as we have already said and it is based on
the signal space concept that we have previously described.
(Refer Slide Time: 01:14)

Suppose we have two orthonormal pulses as discussed in the previous lecture,𝑝1 (𝑡) and
𝑝2 (𝑡).

2
𝑝1 (𝑡) = √ cos(2𝜋𝑓1 𝑡) 0≤𝑡≤𝑇
𝑇

2
𝑝2 (𝑡) = √ cos(2𝜋𝑓2 𝑡) 0≤𝑡≤𝑇
𝑇

Where 𝑓1 = 𝑘1 /𝑇 and 𝑓2 = 𝑘2 /𝑇. 𝑘1 and 𝑘2 being integers.


(Refer Slide Time: 02:07)

(Refer Slide Time: 03:20)

Both 𝑝1 (𝑡) and 𝑝2 (𝑡) have unit energy and are orthogonal to each other.
(Refer Slide Time: 04:38)

Now we take the following digital modulation scheme.

2
𝐴𝑝1 (𝑡) = 𝐴√ cos(2𝜋𝑓1 𝑡) 𝑓𝑜𝑟 𝑏𝑖𝑡 0
𝑇
𝑥(𝑡) =
2
𝐴𝑝2 (𝑡) = 𝐴√ cos(2𝜋𝑓2 𝑡) 𝑓𝑜𝑟 𝑏𝑖𝑡 1
{ 𝑇

(Refer Slide Time: 05:49)


Note here that both 𝑝1 (𝑡) and 𝑝2 (𝑡) differ in frequency.

(Refer Slide Time: 08:08)

One waveform can be obtained from other by shifting the frequency. Hence, this scheme
is known as frequency shift keying or FSK.

(Refer Slide Time: 09:13)

Assuming both 0 and 1 appear with probability half each, we get the average energy per
bit,
1 2 1
𝐴𝑣𝑒𝑟𝑎𝑔𝑒 𝑒𝑛𝑒𝑟𝑔𝑦 𝑝𝑒𝑟 𝑏𝑖𝑡 = 𝐴 𝐸𝑃 + 𝐴2 𝐸𝑃 = 𝐸𝑏
2 2

(Refer Slide Time: 09:57)

Assuming 𝐸𝑝 = 1 we get,

𝐴 = √𝐸𝑏

(Refer Slide Time: 10:40)

Thus, we have the transmitted signal,


2𝐸
√ 𝑏 cos(2𝜋𝑓1 𝑡) 𝑓𝑜𝑟 𝑏𝑖𝑡 0
𝑇
𝑥(𝑡) =
2𝐸
√ 𝑏 cos(2𝜋𝑓1 𝑡) 𝑓𝑜𝑟 𝑏𝑖𝑡 1
{ 𝑇

(Refer Slide Time: 11:28)

At the receiver end we have the received signal,

𝑦(𝑡) = 𝑥(𝑡) + 𝑛(𝑡)

Where n(t) is the additive gaussian white noise


(Refer Slide Time: 11:52)

(Refer Slide Time: 12:48)

Thus, there will be two pulses

𝐴𝑝1 (𝑡) + 𝑛(𝑡)𝑓𝑜𝑟 𝑏𝑖𝑡 0


𝑦(𝑡) = {
𝐴𝑝2 (𝑡) + 𝑛(𝑡)𝑓𝑜𝑟 𝑏𝑖𝑡 1
(Refer Slide Time: 13:39)

Now that we have two types of signal, we need to find out how to match the receive filter.

(Refer Slide Time: 15:01)

For bit 0 we define,

𝑝1 (𝑡) + 𝑝2 (𝑡)
𝑦̃(𝑡) = 𝑦(𝑡) − 𝐴 ( )
2
(Refer Slide Time: 15:41)

𝑝1 (𝑡) − 𝑝2 (𝑡)
= 𝐴( ) + 𝑛(𝑡)
2

= 𝐴 𝑝̃(𝑡) + 𝑛(𝑡)

(Refer Slide Time: 16:26)


(Refer Slide Time: 17:10)

For bit 1 we have,

𝑝1 (𝑡) + 𝑝2 (𝑡)
𝑦̃(𝑡) = 𝑦(𝑡) − 𝐴 ( )
2

(Refer Slide Time: 18:02)

= −𝐴 𝑝̃(𝑡) + 𝑛(𝑡)
(Refer Slide Time: 18:33)

Thus,

𝐴 𝑝̃(𝑡) + 𝑛(𝑡) 𝑓𝑜𝑟 𝑏𝑖𝑡 1


𝑦̃(𝑡) {
−𝐴 𝑝̃(𝑡) + 𝑛(𝑡) 𝑓𝑜𝑟 𝑏𝑖𝑡 0

(Refer Slide Time: 18:58)

This is similar to BPSK, with 𝑝(𝑡) replaced by, 𝑝̃(𝑡)


(Refer Slide Time: 20:15)

(Refer Slide Time: 20:34)

Thus, the receive filter ℎ(𝜏) has to be matched to 𝑝̃(𝑇 − 𝜏),


(Refer Slide Time: 21:32)

The factor of two is simply a constant and will not affect the SNR.

(Refer Slide Time: 22:08)

Hence, we can write,

ℎ(𝜏) = 𝑝1 (𝑇 − 𝜏) − 𝑝2 (𝑇 − 𝜏)
(Refer Slide Time: 22:44)

2 2
= √ cos(2𝜋𝑓1 (𝑇 − 𝜏)) − √ cos(2𝜋𝑓2 (𝑇 − 𝜏))
𝑇 𝑇

Thus, we have derived the receive filter for optimal SMR, in the next module we will
derive the SNR and bit error rate for this scheme.

Thank you very much.

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