Explore topic-wise InterviewSolutions in .

Correct ANSWER is (d) Data compression

The EXPLANATION: Data compression is done by CHECKING redundancy in data. Data redundancy checking is only POSSIBLE when the data is in digital form and hence data compression cannot be done in ANALOG signals.

The correct ANSWER is (d) FAST Fourier transforms

Explanation: The DFT is a complex program that is long and time-consuming to run. In general, COMPUTERS are not fast enough to perform DFT in real time as the signal occurs. Therefore, a special version of the algorithm has been developed to speed up the CALCULATION. Known as the fast Fourier transform (FFT), it permits real-time signal spectrum analysis.

Correct option is (a) True

The best I can explain: With DSP, the filters can have characteristics far superior to those of EQUIVALENT analog filters. Selectivity can be BETTER because of the ease of controlling binary numbers, and the passband or reject band can be customized to the APPLICATION.

Correct option is (C) Data compression

The best I can explain: Data compression is a process that reduces the number of BINARY words needed to REPRESENT a given analog signal. Since analog to digital conversion PRODUCES a HUGE amount of data, for transmission it is a necessity that data is compressed.

The correct answer is (c) High sampling frequency

The BEST I can explain: The time taken for input/output and the processing time together MUST be smaller than the sampling period to ensure the continuous flow of DATA. Since high sampling FREQUENCIES are needed for accurately converting the analog SIGNAL to digital, high speed DSP is a must.

Right choice is (d) Von Neumann bottleneck

For explanation I would say: The accessibility limitation of only one DATA or INSTRUCTION set at a TIME from the memory is called as Von Neumann bottleneck. This has the EFFECT of greatly limiting the execution speed.

The CORRECT choice is (b) Harvard architecture

The explanation is: In a Harvard architecture MICROPROCESSOR, there are TWO memories, a PROGRAM or instruction memory, usually a ROM, and a data memory, which is a RAM. Also, there are two data paths into and out of the CPU between the memories. Because both instructions and data can be accessed simultaneously, very high-speed operation is possible.

Correct option is (a) John Von Neumann

Explanation: Physicist John Von Neumann is GENERALLY credited with creating the stored program concept that is the basis of operation of all digital COMPUTERS. Binary words representing computer instructions are stored sequentially in a memory to FORM a program. The instructions are fetched and EXECUTED one at a time at high speed.

Correct OPTION is (B) Stored in a RAM

To explain: When the signal is converted from analog to digital it is a SEQUENCE of binary NUMBERS which is stored in the RAM. A user defined code that is usually stored in the ROM performs mathematical and other manipulations after which it is converted back into analog signals.

Right choice is (d) PULSE code modulation

To explain: The most widely USED technique for digitizing information signals for electronic data transmission is pulse code modulation. It has uniform transmission quality and also can be used when the SIGNAL traffic is high.

Right CHOICE is (d) PULSE width changes in ACCORDANCE with the amplitude of modulating signal

For EXPLANATION: In PPM, the pulses change position according to the amplitude of the analog signal. The pulses are very narrow. These pulse signals MAY be transmitted in a baseband form, but in most applications, they modulate a high-frequency radio carrier.

Right answer is (a) Pulse amplitude MODULATION

Explanation: Out of all the pulse modulation techniques, Pulse amplitude modulation is the least EXPENSIVE and SIMPLEST to implement. In pulse amplitude modulation, the amplitude of the pulse varies with the amplitude of the MODULATING signal.

Correct answer is (c) 2

Explanation: The amplitude of PWM is binary in nature meaning that it has only two levels. The amplitude of the modulating SIGNALS varies the width of the PULSES GENERATED.

Correct answer is (a) True

To explain I WOULD say: In PULSE modulation, the CARRIER is not TRANSMITTED continuously but in pulses whose width is determined by the amplitude of the modulating SIGNAL. The duty cycle is made in such a way that the carrier is off for a longer time than it bursts hence the average power consumption is low.

Right CHOICE is (d) PULSE FREQUENCY modulation

Explanation: There are four BASIC forms of pulse modulation. They are: pulse amplitude modulation, pulse width modulation, pulse position modulation pulse code modulation. In any form of pulse modulation, the frequency of the signal is not changed.

Right option is (a) Pulse modulation

To explain: In pulse modulation, the information to be transmitted is represented by a series of binary pulses. Since the pulse information is binary in NATURE analog SIGNAL SHAVE to be converted to digital before TRANSMITTING.

The correct answer is (a) 10MHZ

The EXPLANATION is: Operational frequency = 1 / (100 x 10^-9)= 10MHz. Operational FREQUENCIES HIGHER than this would produce errors.

Correct choice is (b) FLASH conversion

Easy explanation: The flash converter PRODUCES an output as fast as the COMPARATORS can switch and the signals can be translated to binary levels by the logic circuits. Comparator SWITCHING and logic PROPAGATION delays are extremely short. Thus, flash converters are the fastest type of A/D converter.

The correct choice is (a) True

For explanation I would say: During its SAMPLING mode, a sample-and-hold (S/H) CIRCUIT ACCEPTS the ANALOG signal and passes it UNCHANGED. In the hold mode, the amplifier remembers a particular voltage level at the instant of sampling.

Correct option is (b) False

The best explanation: SETTLING time is the amount of time it TAKES for the output voltage of a DAC, after a change in binary INPUT, to STABILIZE to within a specific voltage range.The output is thus not an accurate representation of the binary input; it is not usable until it settles down.

Right choice is (d) 13MHZ

For explanation I would say: Frequency of alias SIGNAL = fa = fs – FM = 28-15 = 13MHz.

Correct ANSWER is (a) 1.25V

The best I can EXPLAIN: RESOLUTION = reference VOLTAGE / 2^N = 10/2^3 = 1.25V.

Right CHOICE is (c) Fast microprocessors

The best explanation: Since in undersampling the TIME PERIOD between samples is sufficiently large, this allows slower PROCESSORS to be used for ADC. This ALSO allows low memory capacity for storage and less power consumption.

Right answer is (d) Input amplifiers

To explain: The R-2R Digital to ANALOG converter consists of FOUR MAJOR parts. They are Reference regulators, Resistor network, Electronic SWITCHES, and output amplifiers. An output amplifier is used to SCALE the output voltage of the signal to any desired level.

Correct choice is (d) Flash conversion

The explanation: R-2R, string and weighted CURRENT source CONVERTERS are Digital to analog conversion methods whereas flash conversion is an analog to digital conversion method. Flash CONVERTER uses a voltage ladder along with a COMPARATOR to compare the reference voltage with the input VOLTAGES.

The correct option is (B) 40MHz

For EXPLANATION I would say: According to sampling THEOREM, minimum sampling frequency = 2 x 20MHz = 40MHz.

The correct answer is (a) 4dB

To EXPLAIN I would say: Processing gain(dB) = 10 log[ (fs/2)/BW] = 10 log[ (100/2)/20]= 10 log[ (50/20)] = 4dB.

The correct answer is (b) Improvement in SNR

The explanation: One of the major ADVANTAGES of oversampling is that it increases the signal to noise RATIO(SNR). It decreases the quantization noise by spreading it over a wider frequency. The increase in SNR is called the processing GAIN.

The correct choice is (a) Single stage RC or LC circuits

For EXPLANATION I WOULD say: An antialiasing filter MUST have good selective characteristics. The roll off rate of a single stage RC or LC circuit is too LOW and hence Multistage LC circuits, RC active filter and High order switched capacitor filter circuits are used.

Correct choice is (c) Low pass

To explain: The antialiasing filter is BASICALLY just a low pass filter which allows signals with FREQUENCIES LESS than the CUT off value to pass through. The cut off frequency is usually half of the SAMPLING frequency.

The CORRECT option is (c) Aliasing

To elaborate: Aliasing occurs when a signal is sampled mistakenly at a frequency LESSER than twice the input frequency. It CAUSES a new signal with a frequency of fs-fm to be created near the original signal.

Correct option is (a) True

Best explanation: The modulator acts like a gating circuit that allows the analog WAVE for a particular time period PRODUCING a pulse. The amplitude of the pulse is the INPUT to the ADC which CONVERTS the VOLTAGE into binary numbers.

Right answer is (b) Stairstep wave

Easiest explanation: The DIGITAL to ANALOG converter takes the BINARY number as an input and produces the analog voltage PROPORTIONAL to the binary number. These analog voltages represent specific analog voltage levels and have stairstep characters.

Correct ANSWER is (b) 6mV

Explanation: The minimum VOLTAGE step INPUT is 6/(2^N-1) = 6/1023 =5.865mV ≈6mV. This is the MAXIMUM ERROR that can occur during the conversion process.

Right answer is (a) 255

For EXPLANATION: The number of voltage increments that can be in an N bit system is GIVEN by 2^N-1 = 2^8-1 =255.

Right answer is (d) 70KHZ

Explanation: Fifth harmonic = 5 X signal FREQUENCY = 5 x 14kHz = 70kHz.

Right answer is (d) Quantization error

Easiest explanation: When the NUMBER of bits is not sufficient enough to represent the analog VOLTAGE levels, quantization error occurs. The greater the number of bits, the greater the number of INCREMENTS over the analog range and the smaller the quantizing error.

Correct answer is (d) 12000Hz

The EXPLANATION: Minimum sampling FREQUENCY= NYQUIST frequency = 2x(frequency of ANALOG waveform)

= 2x 6000Hz = 12000Hz.

Right CHOICE is (b) Sampling

The BEST I can explain: An analog signal is continuous and hence contains an INFINITE number of points. The process in which this continuous signal is discretized is called as sampling. The sampling period DETERMINES the ACCURACY of conversion.