Assignment: ECC201A Signals and Systems B.Tech Computer Science and Engineering FET
Assignment: ECC201A Signals and Systems B.Tech Computer Science and Engineering FET
i
Declaration Sheet
Student Name B. PRAVEEN REDDY
Reg. No 17ETCS002043
Programme B.TECH Semester/Year 4TH / 2ND
Course Code ECC201A
Course Title Signals and Systems
Course Date 18/02/19 to 18/03/19
Course Leader Dr T. Christy Bobby
Declaration
The assignment submitted herewith is a result of my own investigations and that I have
conformed to the guidelines against plagiarism as laid out in the Student Handbook. All
sections of the text and results, which have been obtained from other sources, are fully
referenced. I understand that cheating and plagiarism constitute a breach of University
regulations and will be dealt with accordingly.
Signature of the
Date 18/03/19
Student
Submission date
stamp
(by Examination & Assessment
Section)
Signature of the Course Leader and date Signature of the Reviewer and date
ii
Faculty of Engineering and Technology
Assignment – 02
Se Marks
cti Marking Scheme MM First
on ax ark Examiner Moderator
s s Marks
Pa
rt
A1 Essay on DCT for Audio Signal Processing 5
A
Part-A Max Marks 5
Pa
rt
B 1.1 Formulation and solution of the difference equation 6
B
B 1.2 Plotting of traffic 2
B 1.3 Comments on the variability of the traffic 2
B.1 Max 10
Marks
iii
Course Marks Tabulation
B.1
B.2
Marks (Max 25 )
iv
PART A
Discrete Cosine Transform
The Discrete Cosine Transform (DCT) is a transform that is very common when encoding video and
audio tracks on computers. Many "codecs" for movies rely on DCT concepts for compressing and
encoding video files. The DCT can also be used to analyze the spectral components of images as well.
The DCT is very similar to the DFT, except the output values are all real numbers, and the output vector
is approximately twice as long as the DFT output. It expresses a sequence of finite data points in terms
of sum of cosine functions.
Audio Signal Processing
Audio signal processing is at the heart of recording, enhancing, storing and transmitting audio content.
Audio signal processing is used to convert between analog and digital formats, to cut or boost selected
frequency ranges, to remove unwanted noise, to add effects and to obtain many other desired results.
This process can be done on an ordinary PC or laptop, as well as specialized recording equipment.
An audio signal is a representation of sound, typically as an electrical voltage. Audio signals have
frequencies in the audio frequency range of roughly 20 to 20,000 Hz (the limits of human hearing).
Audio signals may be synthesized directly, or may originate at a transducer such as a microphone,
musical instrument pickup, phonograph cartridge, or tape head. Loudspeakers or headphones convert
an electrical audio signal into sound. Digital representations of audio signals exist in a variety of
formats.
Discrete Cosine Transform for Audio Signal Processing
The discrete cosine transform(DCT) signal processing technique come with many applications on a
great number of engineering fields. Here I propose to apply techniques to the compression of audio
signals.
Using spectral analysis and the properties of the DCT, we can treat audio signals as sparse signals in the
frequency domain. This is especially true for sounds representing tones. Here I propose the use of DCT
to obtain an efficient representation of audio signals, especially when they are sparse in the frequency
domain.
By using the DCT as signal preprocessor in order to obtain a sparse representation in the frequency
domain, here I show that the subsequent application of the signals with less information than the well-
known sampling theorem. This means that our results could be the basis for a new compression
method for audio and speech signals.
Conclusion
I have proposed an efficient joint implementation of DCT, as a method to obtain a sparse audio signal
representation, and the application of the compressive sampling algorithm to this sparse signal. The
DCT speech signal representation has the ability to pack input data into as few coefficients as possible.
This allows the quantizer to discard coefficients with relatively small amplitudes without introducing
audio distortion in the reconstructed signal. Although the compressive sampling technique is used
primarily for compression sample images, we achieve reasonable results due to the preprocessing of
the audio signal.
PART B
v
Solution to Question No. B1:
vi
3
Hence, the solution of particular solution is 𝑦 𝑝 (𝑛) = ( ) ∗ (−1)𝑛 --------------(iv)
5
Now, for finding the value of 𝐶1 & 𝐶2 we have to apply the given initial conditions:
Initial conditions are as follow:
𝑦(0) = 𝑑(0), 𝑦(1) = 𝑑(1)
𝑦(0) = 4, 𝑦(1) = 5
Now apply the initial conditions in eq (v) we get the value of 𝐶1 & 𝐶2
3
𝑦(0) = 𝐶1 (1)0 + 𝐶2 (4)0 + ( ) ∗ (−1)0
5
3
𝑦(0) = 𝐶1 + 𝐶2 + 5
3
𝐶1 + 𝐶2 + = 4
5
3
𝐶1 + 𝐶2 = 4 − 5
20−3
𝐶1 + 𝐶2 =
5
17
𝐶1 + 𝐶2 = -------------------(vi)
5
Now, I have to find the value 𝐶1 & 𝐶2 by using the eq (vi) & (vii)
Now, subtract the eq (vi) – eq (vii)
17
𝐶1 + 𝐶2 = 5
28
𝐶1 + 4𝐶2 = 5
- - -
--------------------------
17−28
−3𝐶2 = 5
−11
−𝐶2 = 15
11
𝐶2 = 15
vii
11
Now, put the value of 𝐶2 = in eq (vi)
15
17
𝐶1 + 𝐶2 = 5
11 17
𝐶1 + =
15 5
17 11
𝐶1 = −
5 15
51−11
𝐶1 = 15
40
𝐶1 = 15
8
𝐶1 = 3
Now substitute the value of 𝐶1 & 𝐶2 in the forced equation we get the total solution of the differential
equation:
6
𝑦(𝑛) = 𝐶1 (1)𝑛 + 𝐶2 (4)𝑛 + ( ) ∗ (−1)𝑛
15
8 11 6
𝑦(𝑛) = (3) (1)𝑛 + (15) (4)𝑛 + (15) ∗ (−1)𝑛
Command window:
3.800000
5.200000
14.800000
49.200000
190.800000
753.200000
3006.800000
12017.200000
viii
48062.800000
192241.200000
768958.800000
3075825.200000
12303294.800000
49213169.200000
196852670.800000
787410673.200000
3149642686.800000
12598570737.199999
50394282942.799995
201577131761.199980
806308527038.799930
3225234108145.199700
12900936432574.799000
51603745730289.195000
206414982921150.780000
825659931684593.120000
3302639726738366.500000
13210558906953456.000000
52842235627813816.000000
211368942511255260.000000
845475770045021060.000000
3381903080180084200.000000
13527612320720337000.000000
54110449282881348000.000000
216441797131525390000.000000
865767188526101560000.000000
3463068754104406200000.000000
13852275016417625000000.000000
55409100065670500000000.000000
221636400262682000000000.000000
886545601050728000000000.000000
3546182404202912000000000.000000
14184729616811648000000000.000000
56738918467246592000000000.000000
ix
226955673868986370000000000.000000
907822695475945470000000000.000000
3631290781903781900000000000.000000
14525163127615128000000000000.000000
58100652510460510000000000000.000000
232402610041842040000000000000.000000
929610440167368160000000000000.000000
GRAPH:
29
x 10 Plot of the discrete signal
10
7
value of discrete signals
0
0 5 10 15 20 25 30 35 40 45 50
value of n = 1..50
x
Question No. B2
xi
∞
𝑒 −𝑠𝑡
𝐻(𝑠) = 𝑐 ∗ [ −𝑠 ]
0
𝑐
𝐻(𝑠) = [𝑒 −∞ − 𝑒 0]
−𝑠
𝑐
𝐻(𝑠) = 𝑠 [Now, replace the c by 9 in next line we get,]
8
𝐻(𝑠) = 𝑠 --------(ii)
8
Hence, the Laplace transform of 𝐻(𝑠) =
𝑠
xii
𝑌(𝑠) = 𝐻(𝑠) ∗ 𝑀(𝑠) + 𝐻(𝑠) ∗ 𝑊(𝑠)
Now, put the value of 𝐻(𝑠), 𝑀(𝑠)&𝑊(𝑠) we get the value of 𝑌(𝑠)as
8 1 8 1
𝑌(𝑠) = 𝑠 ∗ −(𝑠+1) + 𝑠 ∗ 𝑠+4
8 1 1
𝑌(𝑠) = [ + ]
𝑠 −(𝑠+1) 𝑠+4
8 (𝑠+1)−(𝑠+4)
𝑌(𝑠) = 𝑠 [ (𝑠+1)(𝑠+4) ]
8 −3
𝑌(𝑠) = 𝑠 [(𝑠+1)(𝑠+4)]
−24
𝑌(𝑠) = [𝑠∗(𝑠+1)∗(𝑠+4)] ----------(v)
−24
Hence, we get the value of 𝑌(𝑠) = [ ]
𝑠∗(𝑠+1)∗(𝑠+4)
Now, replace the value of 𝑌(𝑠), 𝑉(𝑠)&𝐻(𝑠) from eq (vii) , (v) & (ii) we get,
xiii
−24 8
( )−( ) 8 8
𝑠∗(𝑠+1)∗(𝑠+4) 𝑠∗(𝑠+4)
𝑀(𝑠) = 8 [take 𝑠 common then the 𝑠 get divided and we get the eq as]
𝑠
−3 1
𝑀(𝑠) = ((𝑠+1)∗(𝑠+4)) − ((𝑠+4))
−3−𝑠−1
𝑀(𝑠) = (𝑠+1)∗(𝑠+4)
(𝑠+4)
𝑀(𝑠) = − (𝑠+1)∗(𝑠+4)
1
𝑀(𝑠) = −(𝑠+1)
Now, we have to find the inverse Laplace of 𝑀(𝑠) to get the value of 𝑚(𝑡)
1
𝐿−1 [𝑀(𝑠)] = 𝐿−1 [−(𝑠+1)]
1
𝑚(𝑡) = −𝑒 −1𝑡 [Note: Laplace inverse of 𝐿−1 [−(𝑠+1)] = −𝑒 −1𝑡 ]
Now, here we have to take 𝑠(𝑡) = 𝑤(𝑡) ∗ 𝑚(𝑡) (as per L.T.I system)
Then, From above eq of 𝑦(𝑡) = ℎ(𝑡) ∗ 𝑠(𝑡)
Now, apply Laplace transform we get,
𝐿[𝑦(𝑡)] = 𝐿[ℎ(𝑡) ∗ 𝑤(𝑡) ∗ 𝑚(𝑡)]
𝑌(𝑠) = 𝐻(𝑠) ∗ 𝑊(𝑠) ∗ 𝑀(𝑠)
Now, put the value of 𝐻(𝑠), 𝑊(𝑠) & 𝑀(𝑠) we get,
8 1 1
𝑌(𝑠) = 𝑠 ∗ 𝑠+4 ∗ −(𝑠+1)
−8
𝑌(𝑠) = 𝑠∗(𝑠+4)∗(𝑠+1)
xiv
15