0 Voturi pozitive0 Voturi negative

0 (de) vizualizări39 paginiMCQ for digital signal processing

Apr 28, 2019

© © All Rights Reserved

PDF, TXT sau citiți online pe Scribd

MCQ for digital signal processing

© All Rights Reserved

0 (de) vizualizări

MCQ for digital signal processing

© All Rights Reserved

- dsp6
- DsPIC Filter Design Sell Sheet_51438b
- dsp qb1
- Chandu Dsp Manual
- SPFirst-L11_EV0
- Digital signal processing lab manual
- Jntuh 3-2 Qp's May 2016 - Ece
- TP MODUL 3 PSWD
- Low-cost ECG Wireless With Embedded Fuzzy Diagnostic System
- dafir
- Seventh Semester CSE Syllabi
- Biosignal and Bio Medical Image Processing MATLAB Based Applications - John L. Semmlow
- S5 EC Syllabus
- dsp unit3.doc
- Chapter 14
- Generation_Of_Signals.docx
- Iir filter
- DSP_2nd
- Dsp c44
- IJACSIT_V01.1.3

Sunteți pe pagina 1din 39

net/publication/259294754

CITATIONS READS

0 27,855

1 author:

Mahendra Kumar

Indian Institute of Technology Roorkee

72 PUBLICATIONS 188 CITATIONS

SEE PROFILE

Some of the authors of this publication are also working on these related projects:

All content following this page was uploaded by Mahendra Kumar on 15 December 2013.

Mewar University, Gangrar, Chaittorgarh

Subject: DSP

Unit-1

1. DSP stands?

a) Digital signal processing

b) Discrete signal processing

c) Double signal processor

d) None of the above

2

i

N

2. Given that W e , where N 3 . Then F W N can be computed as F

a) 0

b) 1

c) -1

d) e

2 N

i

N 2

3. Given that W e , where N 3. F W can be computed as F

a) 0

b) 1

c) -1

d) e

4. Determine the convolution sum of two sequences x(n) = {3, 2, 1, 2} and h(n) = {1, 2, 1, 2}

a) y(n) = {3,8,8,12,9,4,4}

b) y(n) = {3,8,3,12,9,4,4}

c) y(n) = {3,8,8,12,9,1,4}

d) y(n) = {3,8,8,1,9,4,4}

5. Sampling theorem:

a) fm<fs

b) fs>fm

c) fs>=2fm

d) fs=2fm

6. Application of Convolution:

a) FIR Filtering

b) Addition

c) Manipulation

d) None of these

a) fs<fm

b) fs<2fm

c)fs=fm

d) all of these

a. Discrete Fourier transform

b. digital function transform

c. digital frequency transform

d. none

a. Fast Fourier transform

b. Fourier function transform

c. Fast frequency transform

d. none

2

i

N

a. W e

2

i

N

b. W e

2

i

N

c. W e

d. none

2

i

N

a. W e

2

i

N

b. W e

2

i

N

c. W e

d. none

a. x (k) = {2, 0, 2, 0}

b. x (k) = {1, 0, 1, 0}

c. x (k) = {2, 0, 1, 0}

d. none

a. 1

b. 0

2

i

N

c. W e

d. none

jwn 0

a. e

b. e jwn0

c. 1

d. none

a. eliminate the redundant calculation and enable to analyze the spectral properties of a signal.

b. enable the redundant calculation and redundant to analyze the spectral properties of a signal.

c. a & b

d. none

16. The relation between DFT and Fourier series coefficients of a periodic sequence is

a. X(K) = Ck/N

b. X(K)= Ck

c. X(K) = NCk

d. X(K)=1/Ck

17. If x(n) ------N pt DFT------ X(K) Then x*(-n, (mod N)) -------------N pt DFT--------

___________

a. X*(-K)

b. X*(K)

c. X(-k)

d. X(K)

18. If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for d xa(t)/dt

a. dΩs/df

b. Ωs

c. Ωs/2

d. 2Ωs

19. If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for xa(2t)

a. 2Ωs

b. Ωs/2

c. Ωs

d. Ωs/4

20. If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for xa2(t)

a. 2Ωs

b. Ωs/2

c. Ωs

d. Ωs/4

21. If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for xa(t)Cos(Ω 0 t)

a. Ωs + 2Ω0

b. Ωs * 2Ω0

c. Ωs /2Ω0

d. Ωs - 2Ω0

22. The minimum sampling frequency for xa(t) is real with Xa(f) non-zero only for 9 KHz < |f| <

12 KHz is

a. 4.5 KHz

b. 6 KHz

c. 9 KHz

d. 12 KHz

23. The minimum sampling frequency for xa(t) is real with Xa(f) non-zero only for 18 KHz < |f| <

22 KHz is

a. 8.8 KHz

b. 9 KHz

c. 11 KHz

d. 17.6 KHz

24. The minimum sampling frequency for xa(t) is complex with X a(f) non-zero only for 30 KHz <

|f| < 35 KHz is

a. 6 KHz

b. 5 KHz

c. 15 KHz

d. 17.5 KHz

25. Find two different continuous-time signals that will produce the sequence x(n) = cos( 0.15

nπ) when sampled with a sampling frequency of 8 KHz.

a. sine(1200πt) and Cos(17200πt)

b. Cos(1200πt) and Sine(17200πt)

c. Cos(1200πt) and Cos(17200πt)

d. Sine(1200πt) and Sine(17200πt)

26. A continuous-time signal xa(t) is known to be uniquely recoverable from its samples xa(nTs)

when Ts = 1 ms. What is the highest frequency in Xa( f )?

a. 500 Hz

b. 1000 Hz

c. 700 Hz

d. 5 KHz

27. Suppose that xa(t) is bandlimited to 8 kHz (that is, Xa( f ) = 0 for |f| > 8000), then what is

the Nyquist rate for xa(t)?

a. 16 KHz

b. 4 KHz

c. 8 KHz

d. 12 KHz

28. Suppose that xa(t) is bandlimited to 8 kHz (that is, Xa( f ) = 0 for |f| > 8000), then what is

the Nyquist rate for xa(t)cos(2π . 1000t)?

a. 16 KHz

b. 4 KHz

c. 18 KHz

d. 5 KHz

28. If a continuous-time filter with an impulse response ha(t) is sampled with a sampling

frequency of fs , what happens to the cutoff frequency wc of the discrete-time filter as fs is

increased?

a. wc increases

b. wc decreases

c. wc remains constant

d. wc depends upon fs

29. A complex bandpass signal xa(t) with Xa(f) nonzero for 10 kHz < f < 12 kHz is sampled at a

sampling rate of 2 kHz. The resulting sequence is x(n) = δ(n), then x a(t) will be

a. xa(t) = (1/2000) (Sine(2000πt)/(πt))ej2π(11000)t

b. xa(t) = (1/2000) (Sine(2000πt)/(πt))e-j2π(11000)t

c. xa(t) = (1/2000) (Cos(2000πt)/(πt))e j2π(11000)t

d. xa(t) = (1/2000) (Cos(2000πt)/(πt))e-j2π(11000)t

30. If the highest frequency in xa(t) is f = 8 kHz, then the minimum sampling frequency for the

bandpass signal ya(t) = xa(t) Cos(Ω0t) if Ω0 = 2π.20.103 will be

a. 56 KHz

b. 64 KHz

c. 16 KHz

d. 32 KHz

a. Digital processing needs pre and post processing devices

b. high cost

c. No memory storage

d. none of above

a. Digital processing needs A/D and D/A converters and associated reconstruction filters

b. high cost

c. No reliable

d. none of above

a. low cost

b. stable

c. reliable

d. all of above

a. predictable

b. repeatable

c. Sharing a single processor among a number of signals by time sharing

d. all of above

35. Advantages of DSP are:

a. low cost

b. repeatable

c. storage of data is very easy

d. all of above

a. Military

b. telecommunication

c. consumer electronics

d. all of above

a. medicine

b. seismology

c. signal filtering

d. all of above

a. overlap save

b. overlap add

c. a & b

d. none of above

39. Correlation

a. It gives a measure of similarity between two data sequences.

b. It gives a measure of dis-similarity between two data sequences

c. a & b

d. none of above

40. Find the response of an FIR filter with impulse response h(n)= {1,2,4} to the input sequence

x(n)={1,2}.

a. y(n)={1,4,8,8}

b. y(n)={1,4,6,6}

c. y(n)={1,2,8,8}

d. none of above

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

Unit-2

1. IIR filters

a) Use feedback

d) possible instability

3. The output of two digital filters can be added. Or, the same effect can be achieved by

a) stopband

b) transition band

c) passband

d) ripple

5. A DSP convolves each discrete sample with four coefficients and they are all equal to 0.25. This

must be a

a) low-pass filter

b) high-pass filter

c) band-pass filter

d) band-stop filter

c) less stopband ripple

a) phase response of a lowpass filter

a) non-symmetrical coefficients

b) Gibb's phenomenon

d) a defective accumulator

13. Two digital filters can be operated in cascade. Or, the same effect can be achieved by

14. A DSP convolves each discrete sample with four coefficients and they are all equal to 0.25. This

must be an

a) IIR filter

b) FIR filter

c) RRR filter

a) lowpass filter

b) highpass filter

c) bandpass filter

d) bandstop filter

a) stopband

b) passband

c) transition band

d) ripple

17. A quantizer operates at a sampling frequency of 16 kHz. What is its Nyquist limit?

a) 4 kHz

b) 8 kHz

c) 16 kHz

d) 32 kHz

a) stopband

b) passband

c) transition band

d) ripple

21. If a linear phase filter has a phase response of 40 degrees at 200 Hz, what will its phase response

be at a frequency of 400 Hz (assuming that both frequencies are in the passband of the filter)?

a) 35 degrees

b) 40 degrees

c) 45 degrees

d) 80 degrees

22. Which of the following is used to alter FIR filter coefficients so they smoothly approach zero at

both ends?

a) rectangular window

b) Blackman window

c) Laplace window

d) Hilbert window

a) a phase reversal

c) a phase discontinuity

d) a phase wrap

24. A DSP convolves each discrete sample with these coefficients: -0.25, -0.25, 1.0, -0.25, and -0.25.

This must be a

a) low-pass filter

b) high-pass filter

c) band-pass filter

d) band-stop filter

a) quantization

b) MAC

c) logarithmic transformation

d) vector calculations

26. For the rectangular window function, the transition width of the main lobe is approximately

(here M is the length of the filter)

a) 4*pi*M

b) pi/4M

c) pi*M/4

d) 4*pi/M

27. For the rectangular window function, the first sidelobe will be __________ dB down the

peak of the main lobe.

a) 12 dB

b) 11 dB

c) 13 dB

d) 14 dB

28. For the rectangular window function, the roll-off will be __________ dB/decade.

a) 25 dB

b) 20 dB

c) 15 dB

d) 10 dB

29. For the hamming window function, the width of the main lobe is approximately (here M is

the length of the filter)

a) 8*pi*M

b) pi/8M

c) pi*M/8

d) 8*pi/M

30. For the hamming window function, the peak of the first sidelobe will be at __________ dB.

a) -40 dB

b) -48 dB

c) -43 dB

d) -45 dB

31. For the hamming window function, the side lobe roll-off will be __________ dB/decade.

a) 25 dB

b) 20 dB

c) 15 dB

d) 10 dB

32. For the hanning window function, the width of the main lobe is approximately (here M is the

length of the filter)

a) 8*pi*M

b) pi/8M

c) pi*M/8

d) 8*pi/M

33. For the hamming window function, the peak of the first sidelobe will be at __________ dB.

a) -35 dB

b) -32 dB

c) -40 dB

d) -43 dB

34. For the Blackmann window function, the width of the main lobe is approximately (here M is

the length of the filter)

a) 12*pi*M

b) pi/8M

c) pi*M/8

d) 12*pi/M

35. For the hamming window function, the peak of the first sidelobe will be at __________ dB.

a) -58 dB

b) -48 dB

c) -45 dB

d) -43 dB

a. Delay a copy of the output signal (by x number of samples), and combine it with the new input

signal.

b. Delay a copy of the input signal (by x number of samples), and combine it with the new output

signal.

c. a & b

d. none of above

a. FIR filters are “finite” there is a specific limit to the number of times that any delayed sample

is added to a new input sample.

b. FIR filters are “finite” there is a specific limit to the number of times that any delayed sample

is added to a new output sample.

c. a & b

d. none of above

38. The output of a filter is a function not only of the input at the present time:

a. but also of previous events.

c. a & b

d. none of above

39. FIR filters have ……., and IIR filters have ……….

c. Zeros, zeros

d. none of above

a. As with any feedback device, create a loop, hence the term infinite.

b. As with any non-feedback device, create a loop, hence the term infinite.

c. As with any feedback device, create a open loop, hence the term infinite.

d. None of above

Answers-Key Unit-2:

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

Unit-3

1. The filter coefficients are stored in:

a. binary registers

b. digital system

c. binary memory

d. none

a. degradation of system performance

b. increase system performance

c. grow power

d. none

a. error

b. noise

c. power

d. none

a. quantization effects in A/D conversion

b. product quantization and coefficient quantization errors in digital filters

c. a & b

d. none.

a. limit cycles in IIR filters

b. product quantization and coefficient quantization errors in digital filters

c. a & b

d. none.

a. finite word length effects in FFTs

b. product quantization and coefficient quantization errors in digital filters

c. limit cycles in IIR filters

d. all of above

a. On the number of bits truncated or rounded bits.

b. On the number of bits rounded bits.

c. On the number of bits truncated bits.

d. all of above.

8. The range for negative truncation error for sign magnitude representation is

a. (2 B 2 L ) T 0

B

b. 0 T (2 2 L)

B

c. (2 2 L) T 0

d. none of above

9. The range for positive truncation error for sign magnitude representation is

a. (2 B 2 L ) T 0

B

b. 0 T (2 2 L)

B

c. (2 2 L) T 0

d. none of above

10. The range for truncation error for two’s complement representation is

a. (2 B 2 L ) T 0

B

b. 0 T (2 2 L)

B

c. (2 2 L) T 0

d. none of above

11. The range for round off error for sign magnitude representation is

a. (2 B 2 L ) / 2 R (2 B 2 L ) / 2

B

b. 0 T (2 2 L)

B

c. (2 2 L) T 0

d. none of above

12. The range for round off error for two’s complement representation is

a. (2 B 2 L ) / 2 R (2 B 2 L ) / 2

B

b. 0 T (2 2 L)

B

c. (2 2 L) T 0

d. none of above

a. DR=6B + 10.8

b. DR=3B + 10.8

c. DR=6B + 1.8

d. none of above

a. DR =-2*logPe(n)

b. DR =-logPe(n)

c. DR =-10*logPe(n)

d. none of above

15. x 2 (n )

n 0

1

a. X(z)X(z 1 )z 1dz

2 jc

1

b. X ( z)z 1dz

2 jc

1

c. X(z)X(z 1 )z 1dz

2 c

d. none of above

b. Y ’(z) = [H ideal(z) + E(z)]

c. Y’(z) = [Hideal(z) X(z)]

d. none of above

d. none of above

c. a & b

d. none of above

19. The effects of limit cycles in first and second order systems were studied by

d. none of above

d. none of above

a. Scaling must be done in such a way that no overflow occurs at the summing point.

b. Scaling must be done in such a way that overflow occurs at the summing point.

c. Scaling must be done in such a way that no underflow occurs at the summing point.

d. none of above

22. The necessary and sufficient condition for preventing overflow in a IIR digital filter.

1

a. X n

h i (k )

k

1

b. X

h i (k )

k

1

c. X

h i (k )

k n

d. none of above

23. The necessary and sufficient condition for preventing overflow in a FIR digital filter.

1

a. X n

h i (k )

k

1

b. X

h i (k )

k n

1

c. X M 1

h i (k )

k 0

d. none of above

a. true

b. false

d. none of above

25. In the second order system, under rounding, the output assumes a cyclic set of values of the

deadband.

a. limit-cycle.

b. band-cycle.

c. dead-cycle.

d. none of above

26. With finite precision the reponse does not converge to the origin but assumes cyclically a set

of values:

a. the limit-cycle.

b. band-cycle.

c. dead-cycle.

d. none of above

a. origin.

b. center.

c. mid.

d. none of above

d. none of above

a. Quantisation error in rounding.

d. none of above

d. none of above

31. Below figure shows:

d. none of above

c. a & b

d. none of above

33. A digital system is characterized by the difference equation y(n)=0.9 y(n-1) + x(n) with

x(n)=0 and initial condition y(-1)=12. Determine deadband of the system.

a. [-5,5]

b. [-3 , 3]

c. [-1,1]

d. none of above

a. not

b. less

c. most

d. none of above

35. FIR filters often require more computation, because you must do ………… for each term in

the impulse response.

a. a multiply-add

b. add

c. multiply

d. all of above

a. constant, not

b. not, constant

c. not, not

d. none of above

37. “Linear Phase” (constant delay), If a filter has a ………….delay, the phase shift of the filter

will be t*w, where t is the time delay, and w the natural frequency

(2*pi*f). a.

a. constant

b. variable

c. equal

d. none of above

38. Non-linear delay, This is the part of the phase shift (in and around the filter’s passband) that

is not modeled by a ………...

a. straight line

b. circle

c. square

d. none of above

2 2B

a. p e ( n )

12

2B

2

b. p e ( n )

2

2 B

c. p e ( n )

12

d. none of above

40. If you don’t want a zero at pi, you can’t use a symmetric ……-length filter. You can use an

antisymmetric even length filter if you want a highpass filter, but then you’ll have a zero at DC.

This means that symmetric high pass filters are of …… length.

a. even , odd

b. odd, even

c. even, even

d. none of above

Answers-Key Unit-3:

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

Unit-4

a. Up-sampler

b. down sampler

c. a & b

d. none of above

a. Up-sampler

b. down sampler

c. a & b

d. none of above

a. Up-sampler

b. down sampler

c. a & b

d. none of above

between two consecutive samples of x[n].

a. zero

b. one

c. two

d. none of above

5. Input-output relation for ………..

x[n /L], n 0, L, 2L,

x u [n ]

0, otherwise

a. Up-sampler

b. down sampler

c. a & b

d. none of above

6. In practice, the zero-valued samples inserted by the up-sampler are replaced with appropriate

nonzero values using some type of filtering process, Process is called…….

a. interpolation

b. decimation

c. a & b

d. none of above

7. ………operation is implemented by keeping every M-th sample of x[n] and removing M-1 in-

between samples to generate y[n].

a. Up-sampling

b. Down-sampling

c. a & b

d. none of above

a. Up-sampler

b. down sampler

c. a & b

d. none of above

9. The up-sampler and the down-sampler are ……..but time-varying discrete-time systems:

a. linear

b. none linear

c. a & b

d. none of above

j

10. A factor-of-2 sampling rate expansion leads to a compression of X(e ) by a factor of 2 and

a 2-fold repetition in the baseband [0, 2 ]. This process is called………

a. imaging

b. sampling

c. decimation

d. none of above

11. A ……..is formed by an interconnection of the up-sampler, the down-sampler, and the

components of an LTI digital filter.

c. a & b

d. none of above

12. An interchange of the positions of the branches in a cascade often can lead to a

computationally ……….realization.

a. efficient

b. non-efficient

d. none of above

13. To implement a ……..in the sampling rate we need to employ a cascade of an up-sampler

and a down-sampler.

a. fractional change

b. constant change

c. variable change

d. none of above

with no change in the input-output relation: y1[n] y 2 [n] if and only if M and L are relatively

…..

a. prime

b. non prime

c. natural number

d. none of above

15. From the sampling theorem it is known that a the sampling rate of a critically sampled

discrete-time signal with a spectrum occupying the full Nyquist range cannot be reduced any

further since such a reduction will introduce………..

a. aliasing

b. quantization

c. error

d. none of above

16. The bandwidth of a critically sampled signal must be reduced by ………filtering before its

sampling rate is reduced by a down-sampler.

a. lowpass

b. highpass

c. a& b

d. none of above

appropriate values for an effective sampling rate………...

a. decrease

b. increase

c. a & b

d. none of above

18. Since up-sampling causes periodic repetition of the basic spectrum, the unwanted images in

the spectra of the up-sampled signal x u [n ] must be removed by using a lowpass filter H(z),

called…………………..

d. all of above

filter, called ……………...

d. all of above

20. In the case of single-rate digital signal processing, IIR digital filters are, in general,

computationally more efficient than equivalent FIR digital filters, and are therefore preferred

where computational ………….needs to be minimized.

a. cost

b. memory

c. speed

d. none of above

Decimation

a. Aliasing Step

b. Anti-Aliasing Step

c. a & b

d. all of above

Interpolation

a. Imaging Step

b. Anti-Imaging Step

c. a & b

d. all of above

low-pass filtering prior to sample removal.

a. aliasing

b. quantization

c. a & b

d. all of above

24. A signal can be restored to a higher sampling frequency by the processes of…………….

b. down sampling and decimation

c. a & b

d. all of above

25. ………….have the property of noise-shaping, which allows the elimination of quantization

noise by low-pass filtering.

a. Delta-Delta quantizers

b. Delta-sigma quantizers

c. a & b

d. all of above

26. Care has to be taken with any feedback system. Feedback coefficients have to remain

below……..

a. 1.0

b. 2.0

c. 1.5

d. all of above

c. more computation

d. all of above

c. less computation

d. all of above

c. Ear

d. all of above

a. feedforward filter

b. feedback filter

c. a & b

d. all of above

b. It increases the depth required for accumulators (mantissa for floating point)

c. a & b

d none of above

32. A ……….is nothing but a way to implement a set of filters, generally strongly

mathematically related, in one operation.

a. filterbank

c. node bank

d. none of above

33. A ………can always be decomposed into a set of individual filters; this is usually a lot more

work that it’s worth, but not always.

a. filterbank

c. node bank

d. none of above

a. filter-bank.

b. transform

c. a & b

d. none of above

a. MAC

b. MAA

c. ADD

d. none of above

achieve the desired thoughput.

a. FPGA technology

b. Nanotechnology

c. MEMS technology

d. none of above

37. The roots of polynomial F(z) define the zeros of the filter. FIRs are also called...........

d. none of above

38. A variation of the direct FIR model is called the transposed FIR filter. It can be constructed

from the direct form FIR filter by

d. all of above

39. The direct form FIR filter needs ................ between the adders to reduce the delay of the

adder tree and to achieve high throughput.

d. all of above

40. The FIR filter with transposed structure has registers between the adders and can achieve

high ........without adding any extra pineline registers.

a. throughput

b. speed

c. memory

d. all of above

Answers-Key Unit-4:

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

- dsp6Încărcat deRudra Kumar Mishra
- DsPIC Filter Design Sell Sheet_51438bÎncărcat deeinkatze
- dsp qb1Încărcat deSudharshanan AH
- Chandu Dsp ManualÎncărcat deSumit Padhi
- SPFirst-L11_EV0Încărcat dekewancam
- Digital signal processing lab manualÎncărcat dedeepanilsaxena
- Jntuh 3-2 Qp's May 2016 - EceÎncărcat deKamisetti Tulasi
- TP MODUL 3 PSWDÎncărcat deSophie Dwivita Evans Anthen
- Low-cost ECG Wireless With Embedded Fuzzy Diagnostic SystemÎncărcat deRaghu Kodi
- dafirÎncărcat deRayyan Riaz
- Seventh Semester CSE SyllabiÎncărcat deSubin Vijayan
- Biosignal and Bio Medical Image Processing MATLAB Based Applications - John L. SemmlowÎncărcat deazharzeb
- S5 EC SyllabusÎncărcat deSrikanth Kodoth
- dsp unit3.docÎncărcat deRajasekhar Atla
- Chapter 14Încărcat deNaga Manickam
- Generation_Of_Signals.docxÎncărcat desinghhv21
- Iir filterÎncărcat deBhaskar K
- DSP_2ndÎncărcat deParamesh Waran
- Dsp c44Încărcat deElektro Kotorra
- IJACSIT_V01.1.3Încărcat deDora Teng
- 52208-mt----advanced digital signal processingÎncărcat deSRINIVASA RAO GANTA
- DSP MCQ's.pdfÎncărcat deMUSHTAREEN HARLAPUR
- 7.2-ZimmerÎncărcat demaryam_rahimizade
- 2 Basics DSP AV Z Filters NoiseÎncărcat devignanaraj
- DSP Question Bank B. TechÎncărcat deSayan Datta
- rptIpPrintNew (2)Încărcat deSushant Sharma
- M.tech. - Embedded_SystemsÎncărcat dePaidi Vijay
- 10.1.1.17.8532.pdfÎncărcat deShafiullaShaik
- 10.1.1.78.7735Încărcat deFawaz Mohammed Al-Jubori
- Complete theory lms.pdfÎncărcat dePrabakaran Balasubramaniam

- U2 L3 Generation of AM Wave Square Law Modulator Switching Modulator1Încărcat dealpha_numeric
- Comm 03 digital ModulationÎncărcat deZainab Faydh
- encoding.pdfÎncărcat deZainab Faydh
- Fiber Optics QuestionnaireÎncărcat deLorenz Ardiente
- Comm 03 Amplitude ModulationÎncărcat deHazem Abu Ramadan
- lesson 3 Amplitude Modulation Transmission.pdfÎncărcat deMoses Kaswa
- DIÎncărcat deZainab Faydh
- Pulse Width ModulationÎncărcat desh_ghas6282
- ModulationÎncărcat deZainab Faydh
- Notes_Coherent and Non Coherent DetectionÎncărcat deZainab Faydh
- ecoc07_ddoofdmÎncărcat deZainab Faydh
- CommunicationÎncărcat deZainab Faydh
- Lecture 1Încărcat deZainab Faydh
- Low Coherence InterferomenterÎncărcat deZainab Faydh
- Physics of Wav Group 8-09-10Încărcat deZainab Faydh
- Component DescriptionÎncărcat deZainab Faydh
- [11]Digital Filters for Coherent Optical ReceiversÎncărcat deZainab Faydh
- 05032726Încărcat deZainab Faydh
- SignalÎncărcat deArun
- Part 2Încărcat deZainab Faydh
- IOÎncărcat deattiakhanum

- 18766516-Tcsm3iÎncărcat desinasabikona
- ITU-R BT.656-5Încărcat denikkiauburger
- SE7(17)058_Draft ECC Report on LTE in 400 MHz RevisedÎncărcat deAnonymous 0WSfO04QVb
- Adc BasicsÎncărcat deAmit Tripathi
- 3GPP Long Term Evolution (LTE)Information Block 2 (SIB2) in LTEÎncărcat deSagar Pranith T
- Compact Size of Microstrip Patch Antenna With Slotted Design at 900mhz-Ijaerdv04i0199472Încărcat deEditor IJAERD
- Datasheet Tea5757Încărcat deRubens Vaz Pinto
- EEE C424Încărcat deAnkit Tendulkar
- LifiÎncărcat devamsy
- Measurement of UHF RFID Tag Antenna ImpedanceÎncărcat deMar Cel
- Mivec FaultÎncărcat debjr6627
- Go Az494cmÎncărcat deAlainbravopaez
- Regulador de Voltaje DigitalÎncărcat dejolupeco44
- Wireless Communication Question BankÎncărcat deDeepak Rout
- FlexiHopper DatasheetÎncărcat deslavun
- 00425657-ODU Hardware Description(V100R002_02)Încărcat deMunimbkr
- ZSC-3-2+Încărcat deSagar Prabhakar
- Important Questions on Communication SystemÎncărcat dekeerthy
- GVF White Paper on Licensing Satellite Services in the Ka BandsÎncărcat deradam
- Earphone Headphone Brochure EnglishÎncărcat deCristian Perez Espinosa
- HT0740Încărcat desmhb
- Jugs Radar Wireless UmÎncărcat dealzatex
- Ahmed Abouzeid Project Manager Resume 10 2016 18Încărcat deAhmed El-Osaily
- Celltech Brochure SAR Mar2012 PDFÎncărcat deRebecca Paxton
- 1111Încărcat deapi-19755952
- Lab8_F12Încărcat deEr Satpal Singh Dhillon
- TK-380 Service ManualÎncărcat deDick Einstein
- Mini ProjectÎncărcat deAnonymous VASS3z0wTH
- Eighth International Conference on Wireless, Mobile Network & Applications ( WiMoA 2016 )Încărcat deCS & IT
- pc4216_imÎncărcat deandrei

## Mult mai mult decât documente.

Descoperiți tot ce are Scribd de oferit, inclusiv cărți și cărți audio de la editori majori.

Anulați oricând.