This document describes an experiment to verify Nyquist's criterion for analog signal sampling and reconstruction. It discusses sampling theory, the Nyquist rate, sampling amplifiers, sample and hold amplifiers, and the effects of varying sampling duty cycle and filter order on reconstructed signals. The experiment aims to study these concepts and compare sampling amplifier output to sample and hold amplifier output. Key findings are that signals can only be reconstructed without distortion when sampled above the Nyquist rate, and sample and hold amplification yields better reconstruction than sampling alone.
This document describes an experiment to verify Nyquist's criterion for analog signal sampling and reconstruction. It discusses sampling theory, the Nyquist rate, sampling amplifiers, sample and hold amplifiers, and the effects of varying sampling duty cycle and filter order on reconstructed signals. The experiment aims to study these concepts and compare sampling amplifier output to sample and hold amplifier output. Key findings are that signals can only be reconstructed without distortion when sampled above the Nyquist rate, and sample and hold amplification yields better reconstruction than sampling alone.
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This document describes an experiment to verify Nyquist's criterion for analog signal sampling and reconstruction. It discusses sampling theory, the Nyquist rate, sampling amplifiers, sample and hold amplifiers, and the effects of varying sampling duty cycle and filter order on reconstructed signals. The experiment aims to study these concepts and compare sampling amplifier output to sample and hold amplifier output. Key findings are that signals can only be reconstructed without distortion when sampled above the Nyquist rate, and sample and hold amplification yields better reconstruction than sampling alone.
Copyright:
Attribution Non-Commercial (BY-NC)
Available Formats
Download as TXT, PDF, TXT or read online from Scribd
AIM: 1. To experimentally verify Nyquist criterian for Analog signal sampling and its reconstruction. 2. To study the effect on the reconstructed signal by varying the duty cycle of the sampling signals. 3. To study the difference between the sampling amplifier and sample and hold amplifier output and its effects on reconstruction. 4. To compare the performance of Low pass filter by varying the order of the filter. APPARATUS REQUIRED: 1. Analog signal sampling and reconstruction kit. 2. CRO 3. Power supply 4. Patch cords & wires THEORY: Sampling: It is the process in which an analog signal is converted in to a corresponding sequence of samples that are uniformly spaced in time. To perform sampling, the signal should be band limited and it must obey the Nyquist criterian of sampling theorem. Nyquist criterian: It states that, as long as the sampling rate is at least equal to twice the maximum signal frequency, the signal can be faithfully reconstructed from the samples without aliasing. Sampling is achieved by an electronic switch that switches ON & OFF at the various sampling rates. The samples of the analog signal is being transmitted during the sampling times. This is illustrated in the in the fig 1. The analog switch switches at the rate of the sampling rate selected (2, 4, 8, 16&32 KHz). These sampling clock acts as the control input to the analog switch. Whenever the switch is “ON” ie Ton of the sampling signal, the base band signal from the unity gain amplifier is latched to the output. At reconstruction side, a LPF gathers all the samples to reconstruct the original source signal. 2/129 SAMPLING SIGNAL (2,6,8,16,32 KHZ) Signal I/P Sampled O/P CD 4016 Analog Switch SAMPLING LOGIC + - + - C Input g(t) Out Put u(t) Fig. a Out Put u(t) Input g(t) Fig. b SAMPLE AND HOLD AMPLIFIER Amplifier 3/129 4 3 2 1 Switch Positions Duty Cycle 4 3 2 1 10% ON ON ON ON 20% ON ON ON OFF 30% ON ON OFF ON 40% ON ON OFF OFF 50% ON OFF ON ON 60% ON OFF ON OFF 70% ON OFF OFF ON 80% ON OFF OFF OFF 90% OFF ON ON ON Sample and Hold: The sample and hold circuit is used to improve the signal power at the out put of the low pass filter in the receiver. In its ideal form, the sample and hold circuit produces an output waveform that represents a staircase interpolation of the original signal. In sampling the spectrum of the sampled signal is scaled by the ratio T/Ts, where T is the sampling pulse duration and Ts is the sampling period. Typically this ratio is very small, with the result that the signal power at the output of the LPF in the receiver is correspondingly small. We may obviously overcome this situation by use of Amplification, which may be quite large. Using a Sample and hold circuit provides better remedy to overcome this situation. Fig 2 shows the typical Sample and hold circuit. Here analog switch is timed to close only for the small duration of T of each sampling pulse, during which time the capacitor rapidly charges up to a voltage level equal to that of the input sample. When the switch is open, the capacitor retains its voltage level until next closer of the switch. Thus the sample and hold circuit produces an output wave form that represents a staircase interpolation of the original signal. 4/129 VERIFICATION OF NYQUIST RATE The input analog Signal at 2 Khz 32 KHz Sampling and Sampled output 16 KHz Sampling and sampled output 8 KHz Sampling and Sampled output 4 KHz Sampling and sampled output 2 KHz Sampling and sampled output 5/129 Effects of Duty cycle: The sampling theorem does not take into account of duty cycle as long as enough number of samples of information (i.e. at least the Nyquist rate) are being transmitted. Since enough information at frequent regular intervals of time are being transmitted, there are no problems in recovering the information from the samples irrespective of the duty ratio of the sampling signal. However when the sampling frequency is set equal to base band signal frequency with the duty cycle of the sampling frequency set to nearly 90% , the signal can be reconstructed once again faithfully. PROCEDURE 1. VERIFICATION OF NYQUIST RATE • Connect 2 KHz, 5v Pk-Pk signal generated on board to the ANALOG INPUT by patch cords • Connect the sampling frequency signal in the INTERNAL mode by means of the shorting jumper setting provided in the kit itself. • Observe the out put of the sampling amplifier at the SAMPLE OUTPUT • Connect the SAMPLE OUTPUT to the input of the 2nd order, 4th order& 6th Order LPF. • Vary the sampling frequency from 2 KHz to 32 KHz by pressing FR SEL and observe the reconstructed signal each time. Particularly verify the Nyquist criterian at the sampling frequency of 2 KHz ie at the same frequency level of base band signal & 4 KHz. 2. STUDYING THE EFFECT OF DUTY CYCLE • Connect 2 KHz, 5v Pk-Pk signal generated on board to the ANALOG INPUT by patch cords • Connect the sampling frequency signal in the INTERNAL mode by means of the shorting jumper pin provided in the kit itself. • By means of the DIP switch setting, as indicated in the duty cycle table vary the duty cycle of the sampling frequency signal from 10% to 90% in the discrete step of 10% each. • Observe the effect of duty cycle at INTERNAL SAMPLING FREQENCY. • Observe the effect of duty cycle on sampling amplifier at SAMPLE OUTPUT. 6/129 EFFECT OF VARYING THE DUTY CYCLE 20% Duty Cycle Sampling frequency at 4 KHz 30% Duty Cycle Sampling frequency at 4 KHz And reconstructed output at 4th order filter And reconstructed output at 4th order filter 50% Duty Cycle Sampling frequency at 4 KHz 70% Duty Cycle Sampling frequency at 4 KHz And reconstructed output at 4th order filter And reconstructed output at 4th order filter 7/129 • Vary the sampling frequency from 2 KHz to 32 KHz by pressing FR SEL and observe the reconstructed signal output across 2nd, 4th & 6th order LPF. Particularly verify the reconstructed signal at the output of 2nd & 4th order filter for the sampling frequency of 2 KHz with Duty cycle of 90%. 3. SAMPLING VS SAMPLE AND HOLD • Connect 2 KHz, 5v Pk-Pk signal generated on board to the ANALOG INPUT by patch cords • Connect the sampling frequency signal in the INTERNAL mode by means of the shorting jumper pin provided in the kit itself. • Vary the sampling frequency from 2 KHz to 32 KHz by pressing FR SEL and observe the sampling amplifier at SAMPLE OUTPUT and reconstructed signal at the output of 4th order LPF. • Vary the sampling frequency from 2 KHz to 32 KHz by pressing FR SEL and observe the sample and Hold amplifier at SAMPLE HOLD OUTPUT and reconstructed signal at the output of 4th order LPF. • Compare the signals of SAMPLE OUTPUT and SAMPLE AND HOLD OUTPUT & also the reconstructed signal output observed for various sampling frequencies. 4. EFFECT OF VARYING THE ORDER OF THE FILTER • Connect 2 KHz, 5v Pk-Pk signal generated on board to the ANALOG INPUT by patch cords. • Connect the sampling frequency signal in the INTERNAL mode by means of the shorting jumper pin provided in the kit itself. • Connect SAMPLE &HOLD OUTPUT to the input of the 2nd order LPF &observe the reconstructed signal at the output of the 2nd order LPF. • Connect SAMPLE &HOLD OUTPUT to the input of the 4th order LPF &observe the reconstructed signal at the output of the 4th order LPF. • Connect SAMPLE &HOLD OUTPUT to the input of the 6th order LPF &observe the reconstructed signal at the output of the 6th order LPF. • Select the duty cycle of 10% & observe the reconstructed signal output for 2nd, 4th & 6th order LPF. 8/129 SAMPLING Vs SAMPLE & HOLD Sampling 32 Khz and output at 2nd order LPF Sampling 32 Khz and output at 4th order LPF Sampling 16 Khz and output at 2nd order LPF Sampling 16 Khz and output at 4th order LPF Sampling 8 Khz and output at 2nd order LPF Sampling 8 Khz and output at 4th order LPF Sampling 2 Khz and sampled only and Sampling 2 Khz and sample & Hold and output at 4th order LPF output at 4th order LPF 9/129 INFERENCES: 1. VERIFICATION OF NYQUIST RATE : We observe that by sampling a 2 KHz base band signal at Nyquist rate of 4 KHz and above , the signal is reconstructed. But by sampling the same 2 KHz Base band signal at 2 KHz (< Nyquist rate), we observe that there is a distortion in the reconstructed signal. From the above we infer that, for signal reconstruction with no distortion, Nyquist criterian has to be satisfied. If Nyquist criteria is not satisfied, or if the signal is not band limited, then the spectral overlapping & aliasing occurs, causing high frequency components show up at the low frequencies in the recovered signal. 2. EFFECT OF DUTY CYCLE: We observed that varying the duty cycle has no effect as long as the Nyquist criterian is satisfied. However when the sampling frequency is set to 2 KHz which is same as that of the base band signal with the duty cycle of the sampling frequency set to 90%, the signal is reconstructed once again faithfully, even though Nyquist criterian is not satisfied. This is because, during this Ton period of 90%, most of the information being conveyed even in a single sample. 3. SAMPLING VS SAMPLE AND HOLD: From the observations, we conclude that the signal is reconstructed more precisely in the case of Sample & Hold amplifier rather in case of sampling amplifier, where the reconstructed signal suffers an amplitude distortion. This is because the spectrum of the sampled signal is scaled by the smaller ratio of T/Ts. Hence we infer that use of Sample & Hold is an excellent logic which compensates for the poor amplitude at the input of the LPF resulting in faithful reconstruction of the signal. 4. EFFECT OF VARYING THE ORDER OF THE FILTER: We observe that the signal is reconstructed more faithfully in a 4th & 6th order LPF compared to the 2nd order LPF. Especially in the case when the duty cycle is kept at 10%, the signal suffers with distortion at the output of 2nd order LPF. This is because when the order of the filter is low, the rolloff rate decreases, resulting in unfaithful reproduction of original signal. Where as in the case of 4th & 6th order filter, the performance & efficiency is improved due to increased Roll-off rate. 10/129 11/129 RESULT 1. Nyquist criterion for an Analog signal sampling & its reconstruction is being experimentally verified. 2. Experimentally studied the effect on the reconstructed signal by varying the duty cycle of the sampling signals. Found that, as long as Nyquist criterian is satisfied, the duty cycle has no importance. However the signal can still be reconstructed, even though the Nyquist criterian does not holds good in the case of duty cycle being selected to its maximum level (say 90%) 3. Experimentally studied the difference between the sampling amplifier and sample and hold amplifier output and its effects on reconstruction. We found that use of sample & Hold is an excellent logic which compensates for the poor amplitude at the input of the LPF resulting in faithful reconstruction of the signal. 4. Experimentally compared the performance of Low pass filter by varying the order of the filter. We found that the signal is reconstructed more faithfully in the 4th & 6th order LPF compared to the 2nd order LPF. 5. All necessary waveforms gathered from CRO being attached. 12/129 13/129 Time Division Multiplexing AIM: To learn the concept of Time Division Multiplexing technique and transmit four different signals using the same channel. To understand clock recovery techniques using three-wire, two-wire and single – wire methods. APPARATUS REQUIED: TDM KIT, Oscilloscope, Power supply, connecting wires. THEORY: Multiplexing is a kind of signal processing operation whereby, a number of independent signals can be combined into a composite signal suitable for transmission over a common channel. This is accomplished by separating the signals either in frequency or time. � The technique of separating the signals in frequency is referred to as Frequency Division Multiplexing. � The technique of separating the signals in time is referred to as Time Division Multiplexing. CONCEPT OF TDM: The sampling theorem provides the basis for transmitting the information contained in a band-limited message signal m(t) as a sequence of samples of m(t) taken uniformly at a rate that is usually slightly higher than the Nyquist rate. An important feature of sampling process is conservation of time. That is, the transmission of the message samples engages the communication channel only for a fraction of the sampling interval on a periodic basis, and in this way some of the time interval between adjacent samples is cleared for use by other independent message sources on a time shared basis. 14/129 TYPICAL TDM WAVEFORM Multiplexed output at TxD 15/129 We thereby obtain a TDM system, which enables the joint utilization of a common communication channel by a plurality of independent message sources without mutual interference among them. For a given message signal, transmission of the association PAM wave engages the communication channel only for a fraction of the sampling interval between adjacent pulses pf the PAM wave. The channel is cleared for use by other independent message signals on a time shared basis. Each input message signal is first restricted in bandwidth by a low-pass pre-alias filter to remove frequencies that are non essential to an adequate signal representation. The pre-alias filter outputs are then applied to a commutator, which is usually, implemented using electronic switching circuitry. The function of commutator is two fold: � to take a narrow sample of each of the N input messages at a rate fs that is slightly higher than 2W, where W is cutoff frequency of the pre-alias filter, � to sequentially interleave these N samples inside a sampling interval Ts = 1/fs. Following the commutation process, the multiplexed signal is applied to a pulse – amplitude modulator. This is done to transform the multiplexed signal into a form suitable for transmission over the communication channel. At the receiving end, the received signal is applied to a pulse-amplitude demodulator, which performs the reverse operation of the pulse amplitude modulator. The short pulses produced at the pulse demodulator output are distributed to the appropriate low-pass reconstruction filters by means of a decommutator, which operates in synchronism with the commutator in the transmitter. This synchronization is essential for a satisfactory operation of the TDM system. 16/129 SINGLE-WIRE TRANSMISSION 17/129 SYNCHRONIZATION: Synchronization is the most critical aspect of TDM. There are two levels of synchronization in TDM viz., � Frame synchronization � Sample synchronization To maintain proper positions of sample pulses in the multiplexed it is necessary to synchronous the sampling process. Because the sampling operations are usually electronic, there is typically a clock pulse train that serves as a reference for all samples. At the receiving station, there is similar clock that must be synchronized with the transmitter. Clock synchronization can be derived from the received waveforms by observing the pulse sequence over many pulsed and averaging the pulses (in a closed loop with the clock derived on the VCO). Clock synchronization does not guarantee the proper sequence of samples is synchronized. Proper alignment of time slot sequence requires frame synchronization. One or more time slots per frame may be used for sending frame synchronization information. For example, placing a special pulse that has a larger amplitude than the largest expected message amplitude, the start of the frame can be easily identified using a threshold circuit. CIRCUIT DISCRIPTION: The kit basically consists of the following sections: a) ON-BOARD FUNCTION GENERATOR: The unit provides four amplitude variable synchronized sine waves for four channels 250Hz, 500Hz, 1 KHz, 2 kHz. The amplitude of each sine wave can be varied from 0 to 5Vpeakpeak. An amplitude variable DC level (0-5V) is also provided for generation of synchronization pulses. The 32 KHz sampling clock and 8 KHz channel identification signal are also generated by the circuit. The sampling rate is 32 KHz. 18/129 OBSERVED TDM WAVEFORMS At Input Channel 1: Input At Output Channel 1: Receiver Frequency 250 Hz Output Frequency 250 Hz At Input Channel 2: Input At Output Channel 2: Receiver Frequency 500 Hz Output Frequency 500 Hz At Input Channel 3: Input At Output Channel 3: Receiver Frequency 1KHz Output Frequency 1KHz 19/129 b) TRANSMITTER: The transmitter section consists of a four analog input with 4-pole integrated circuit analog switch that provides sampling and TDM of each channel, using PAM signals. b) RECEIVER: The receiver section consists of a 4-pole analog switch that demultiplexes the 4 channels. A 4Th order low pass filter is provided for each receiver channel, in order to obtain the information from the multiplexed and PAM demodulated signals. The cutoff frequency of the 4TH order filter is 3.4 KHz. A PLL circuitry is used to generate the clock and channel information at the receiver. Here the phase comparator used is an edge controlled Digital Memory network. This type of phase comparator acts only on positive edges of the signal and comparator inputs are not important. The comparator threshold level control is provided for synchronous pulse detection, when synchronization pulsed are transmitted on one of the channels. OPERATION OF THE KIT: LEVEL-1: THREE-WIRE TRANSMISSION: The Transmitter, Clock and the Channel identification signals are directly linked with the receiver. Here, in this method three wires are used from Tx to Rx, one for data, one for clock and the other one for channel. Hence the name three wire transmission. PROCEDURE: � The four channels are connected to the multiplexer input channel. � The Tx signals viz., TxD, TxCH0, TxCLOCK are directly connected to their receiver counterparts viz., RxD, RxCH0, and RxCLOCK respectively. � The sampled signals are observed at the transmitter end at the receiver end. INFERENCE: Since the clock is also transmitted to the receiver, reconstruction is easy. Fourth order low pass filters are used to obtain the output. 20/129 OBSERVED TDM WAVEFORMS At Input Channel 4: Input At Output Channel 4: Receiver Frequency 2KHz Output Frequency 2 KHz Multiplexed output at TxD At Input Channel 1: DC signal At Output Channel 1: DC signal 21/129 LEVEL-2: TWO-WIRE TRANSMISSION: In two wire transmission, only the channel identification is linked directly to the receiver, while the RxCH0 and RxCLOCK are generated by the PLL circuitry. PROCEDURE: � The four channels are connected to the multiplexer input. � TxD is connected to RxD, TxCH0 to the PLL circuitry’s input. � (PLL output)- Sync and clock are connected to RxCH0 and RxCLOCK respectively. � The sampled signals are observed at the transmitter end and at the fourth order filter output as reconstructed. INFERENCE: The PLL is used to generate the channel identification info and CLOCK. PLL locks on to the TxCH0 signal frequency and thereby provides synchronization. LEVEL-3: SINGLE-WIRE TRANSMISSION: In single wire transmission only the TXD data is connected to RXD. The receiver derives the clock and channel identification signals from the incoming stream of TDM pulses. Channel 0 sends the sync pulse. RxCLOCK is obtained using the threshold level comparator and PLL. PROCEDURE: � Set the sync level to the maximum amplitude and connect it to the channel 0. � Three of the 4 inputs are given to the remaining channels. Their amplitudes are less than the sync signals. � TxD is connected to RxD. � The PLL input switch is set. � Vary the threshold level comparator so that DC level transmitted is recovered as such in the first channel. � The PLL circuit is used to generate RxCH0 and RxCLOCK signals. 22/129 23/129 INFERENCE: In all three transmission methods, the receiver output is got to be the same. In single wire transmission, the number of transmission lines is reduced to one but the number of inputs is also reduced by one. The PLL locks on to the frequency of sync pulses thereby synchronizing the transmitter to the receiver. RESULT: From the above experiments, it is clear that, regardless of the type of transmission used, the receiver output is got to be the same. Since single wire transmission uses only one wire to connect from Tx to Rx, Single wire transmission proves to be more economical. Thus in TDM, the different message sources are multiplexed in maintaining the individuality of each message throughout. As the number of independent message sources is increased, the time interval allotted has to be reduced. If pulses become too short, impairments in the transmission medium begin to interfere with the operation of the system. Accordingly, in practice, it is necessary to restrict the number of independent message sources that can be included in a Time Division Group. 24/129 25/129 AMPLITUDE MODULATION AND DEMODULATION AIM To generate and observe Amplitude Modulated Signal To vary and set the required Percentage of Modulation To demodulate the AM signal using i) Envelope detector ii) Square Law detector iii) Coherent Demodulation APPARATUS REQUIRED Experiment Kit, Cathode Ray oscilloscope, Power supplies, Connecting wires, Probes. THEORY If Am cos (2πfmt) is the message signal (audio) and Ac cos (2πfct) is the carrier, then m, the percentage of modulation is, Let the sinusoidal carrier wave be defined as c (t) = Ac cos(2πfct) Where Ac – Carrier Amplitude fc – Carrier Frequency If m(t) = Am cos(2πfmt) and m-percentage of modulation The modulated wave is S(t) = Ac[1 + m cos(2πfmt] cos(2πfct) 26/129 27/129 % of Modulation = Amax - Amin Amax + Amin Where Amax – maximum amplitude of s(t) Amin – maximum amplitude of s(t) Total power in the AM wave is Pt = (1 + m2 /2)Pc Where Pc – Carrier Power PROCEDURE GENERATION OF AM 1) The Message signal is connected to the AM input stage. 2) The output of AM input is connected to the multiplier, which has carrier wave as another input. 3) The output of the multiplier is the AM signal. DEMODULATION OF AM ENVELOPE DETECTOR Envelope detector is a non-linear device which consists of a diode and resistorcapacitor filter. On a positive half-cycle of the input signal, the diode is forward biased and the capacitor charges up rapidly to the peak value of the input signal. When the input falls below this value, the diode becomes reverse-biased and the capacitor discharges slowly through the load resistor. The discharging process continues until the next positive half cycle. Thereafter, the charging-discharging routine is continued. PROCEDURE: Connect CH1 to AM input stage and CH2 to output of multiplier. Now both modulating tone and AM signals are displayed on screen. Observing AM, slowly vary the potentiometer 28/129 Analog signal and Carrier Carrier Frequency response (FFT) Over Modulation Under Modulation Critical Modulation Modulated signal Frequency Response (FFT) 29/129 connected to AM input stage such that modulation varies from zero to cent percentage and more. Set m=0.5 and put the oscilloscope in X-Y mode to get the trapezoidal figure as Now set m = 100% and obtain XY plot Make sure noise generator is off. AM signal is connected to envelope detector via noise adder. Observe output of envelope detector in CH1. Now keeping m=1, switch ON noise generator, the noise output level is -40dB. Observe the LPF output on CRO. Reduce the noise level in steps of 1dB and find the threshold when waveform loses its shape. At this threshold, measure signal voltage and noise voltage by connecting true RMS voltmeter to the signal and noise inputs of noise adder. SQUARE LAW DETECTOR A square-law detector simply means that the DC component of diode output is proportional to the square of the AC input voltage. So if you reduce RF input voltage by half, you'll get one quarter as much DC output. Or if you apply ten times as much RF input, you get 100 times as much DC output as you did before. An increase of 3dB results in twice as much output voltage. PROCEDURE The incoming signal is fed to both the inputs of a multiplier. The output signal is passed through a low pass filter. The output is the message. Mathematical Representations: Incoming, x (t) = Ac [1 + m cos (2πfmt)] cos(2πfct)+ Noise Output of multiplier = [ x(t) ]2 = [Ac [1 + m cos(2πfmt)] cos(2πfct) + Noise]2 = Ac 2 m cos(2πfmt) + Higher frequency terms Output of low pass filter = Ac 2 m cos(2πfmt) Thus, the message signal has been retrieved from the incoming modulated signal. m = (A-B)/(A+B) 30/129 Percentage of Modulation 31/129 STUDY EFFECT OF NOISE Now keeping m = 1, Switch ON the Noise Generator; the noise output level is –40dB. The noise passes through a BPF so that it is band limited to that of signal bandwidth. Noise Adder adds this noise with the signal. Observe the LPF output on the CRO. You should now be able to see slight distortion in that received signal. Increase the Noise level in steps of 1dB (ie. -39dB, - 38dB…) by pressing UP button. You could see that the demodulated signal is getting more distorted as you increase the noise level. Also you could hear the 3 KHz tone is distorted due to noise. As the noise level crosses the threshold, the displayed waveform is distorted beyond recognition and the speaker output is unintelligible. Reduce the noise level, by pressing DOWN button, so that the displayed waveform is only slightly distorted. Now adjust the noise level for threshold. That is, increase the noise level by visually observing the waveform and aurally listening to the tone. At some point of time you could see that the 3 KHz tone losses its wave shape and also you have difficulty in listening to the tone clearly. The loss of message in an envelope detector that operates at a low signal-to-noise ratio is referred as the threshold effect. Threshold means a value of the signal-to-noise ratio below which the noise performance of a detector deteriorates much more rapidly than proportionately to the signal-to- noise ratio. SIGNAL TO NOISE RATIO (SNR) It is defined as, (SNR)o = Average power of message signal at the receiver output Average power of noise at the receiver output Output signal to noise ratio is used to measure fidelity of the received message. As long as the noise and demodulator output are additive, output signal to noise ratio is good measure. Output signal at noise ratio depends upon types of modulation and demodulation. 32/129 33/129 INFERENCE Thus from the waveform we obtain that when m tends to a value grater than 1 (over modulation) we get distortion at the output. The non-linearity present in diode restricts envelope detector being used for demodulation large amplitude signals. Comparison of Envelope Detector and Square Law Detector Envelope Detector Square Law Detector 1. Don’t Square the Signal 1. Square the Signal 2. High Dynamic Range 2. Less Dynamic Range 3. It works only when |Ka m(t)| < 1 3. It works for any Ka m(t) 4. Poor SNR 4. Good SNR 5. No Distortion 5. Less Distortion SPECTRUM OF AM Analysis shows fig.1 that the sidebands of the AM, when derived from a message of frequency μ rad/s, are located either side of the carrier frequency, spaced from it by μ rad/s. As the analysis predicts, even when m>1, there is no widening of the spectrum. This assumes linear operation; that is, that there is no hardware overload. The above fig.2 shows the FFT plot of spectrum of any signal. RESULT Thus the AM signal has been generated with varying percentages of modulation. The demodulation of the AM signal has been performed using an envelope detector and a square law detector. 34/129 Block diagram of a method for generating a Narrowband FM Phasor Diagram of narrowband FM Block diagram of the indirect method of generating a wideband FM signal 35/129 Frequency Modulation and demodulation Objectives To generate narrow band and wide band FM signal To demodulate FM signal using Phase Locked Loop (PLL) technique To Calculate Signal to Noise ratio To analyze FM spectra using Bessel functions Requisites 1. Aces experimenter kit for FM modulation and Demodulation. 2. CRO 3. Connection Wires and Jacks 1 Introduction AM uses the amplitude of the carrier to transmit the information. Since the amplitude of the AM modulated waveform is directly proportional to the amplitude of the message, AM modulation is a linear operation. Another way to transmit the information is Frequency Modulation (FM). In this case, it is no longer the amplitude of the carrier, but its instantaneous frequency, that is changed according to the variations of the message signal's amplitude. FM belongs to the family of Angle Modulation methods. Phase Modulation is also an Angle Modulation method, but will not be studied in this Lab. Angle modulation methods are nonlinear operations. It turns out that an FM system uses more bandwidth than an AM system. As bandwidth is a very limited resource in communication systems, we can therefore question the use of FM. You will discover during this Lab that the reason why FM systems are widely used is that this excess of bandwidth allows them to have a much better immunity to noise than AM systems. 2 Theories about FM 2.1 Definition of an FM signal 36/129 Plot of Bessel Functions of first kind for varying Order Number of significant side band frequencies of wideband FM signal for varying modulation index 37/129 2.2 The single tone message example 2.3 Narrow-Band and Wide-Band FM 2.3.1 Narrow-band FM 2.3.2 Wide-band FM 38/129 The above figure shows AM and FM signals produced by a single tone. a.) Carrier wave. b.) Sinusoidal Modulating signal. c.) Amplitude modulated signal. d.) Frequency modulated signal 39/129 40/129 Signal flow diagram for FM generation VCO Voltage Controlled Oscillator 3KZ 6KZ 120KZ FM I/P STAGE FM 41/129 42/129 Band Pass Filter Noise Adder Phase Detector Loop Filter Voltage controlled Oscillator Power AMP Low Pass Filter Signal Flow Diagram for FM Demodulator Phase Lock Loop Noise in FM receiver a.) Power spectral density of quadrature component nQ(t) of narrowband noise n(t). b.)Power spectral density of noise nd(t) at the discriminator output. C.) power spectral density of noise no(t) at the receiver output Divide by N Counter Noise Output FM 43/129 3 The Characteristics of the FM wave: 1. The Zero crossings of FM wave no longer have a perfect regularity in their spacing; Zero crossing refer to instants of time at which a wave form changes from a negative to positive value or vice versa. 2. The envelope of FM wave is constant and equal to the carrier amplitude. 3. The average power of FM wave is constant and equal to ½ A2 c. Where Ac is the amplitude of the carrier wave. 4. The Frequency deviation, representing the maximum departure of the instantaneous frequency of the FM wave from the carrier frequency fc is Δf = Kf Am. Am is the amplitude of the modulation base band signal m(t). 5. The modulation index of the FM wave, β is defined as the ratio of the frequency deviation Δf to the modulation frequency fm and is written as β = Δf /fm 6. The transmission bandwidth of FM wave by Carson’s rule is BWT = 2( β+1)fm 7. β is small (usually 0.5) for narrow-band FM and is large for wide-band FM. 8. The transmission bandwidth of a narrow-band FM wave is closely equal to twice the message bandwidth, whereas in the case of wide-band Fm wave it is well in excess of the value. 4 Practical FM Modulation Procedure: 1. Connect CH1 of oscilloscope to the out put of VCO. 2. Disconnect the Input to the VCO and adjust potentiometers connected to it so that the VCO frequency displayed on the scope is 120 kHz. 3. Connect the CVO Input back. Don't disturb the VCO potentiometer. 4. Now the Input of the VCO frequency modulated signal the Frequency deviation Δf is adjusted by the potentiometer connected to FM Input stage 44/129 Discrete amplitude spectra of an FM signal normalized with respect to the carrier amplitude, for the case of sinusoidal modulation of varying frequency and fixed amplitude. Only the spectra for positive frequencies are shown 45/129 Narrow – Band FM Generation: 1. Adjust the potentiometer connected to FM Input stage to minimum so that Δf = 0. 2. Only the carrier Frequency fc of 120khz is desplayed on the screen. Next connect CH1 to the output of FM Input stage. 3. Set the Volt/Div of scope to 10mv slowly vary the potentiometer connected to the FM Input stage such that the Out put is 20mv peak to peak. 4. Next connect CH1, to the Output of VCO. The displayed wave form is the narrow Band FM. Now the β < = 0.5 Wide – Band Generation: 1. Switch off the Noise generator. Set the Volt/div of CH1 to 100mv 2. Connect CH1 to the Output of FM Input stage. 3. Slowly vary the potentiometer connected to the FM Input stage such that the Output of it is 250 mv peak to peak. 4. Next connect CH1 to the Output of VCO. The displayed waveform is wide- band FM. Now the 2 < β < 3. FM Demodulation Using PLL: The FM wave is added with the band limited noise and feed to the Input of PLL. The PLL demodulates the FM. The PLL demodulate Out put is given to a low pass filter to remove the high frequency carrier residues the noise. EFFECT OF NOISE: When increase the Input noise, the Signal – to – Noise Ratio (SNR) reduces. At low SNR, the FM receiver breaks. At first, individual clicks are heard in the receiver output due to the occurrences of spikes (Spike noise or Impulse noise ) in otherwise smooth white noise. The number of clicks heard in the speaker is proportional to the number of spikes appearing in the demodulated output As the SNR decreases further, the individual clicks rapidly merge in to 46/129 Carrier with Frequency response (FFT) Carrier and modulating signal Narrow Band Fm with Frequency response (FFT) Narrow band FM with Modulating signal Wide Band Fm with Frequency response (FFT) Wide band FM with modulating signal 47/129 crackling or sputtering sound. This indicates the receiver breakdown. For all practical purposes, the occurrences of spikes in de-modulated signal indicate the threshold. Threshold means a value of the signal-to-noise ratio below which the noise performance of a detector deteriorates much more rapidly than proportionately to the signal-to- noise ratio. BESSEL FUNCTION: Noise Performance at J0 = 0 Switch On the Noise Generator. The default noise level is -40dB Connect Ch1 to the Output of Low pass filter. Observing the demodulated output, increase the noise level slowly in steps to 1 dB. As the noise level increases, the waveform shows distortion due to noise and smooth noise along with the 3 KHz tone could be heard form the speaker. If noise level is further increased, spikes appear in the waveform. This is the point of threshold. If the noise level is further increase, the receiver breaks down and crackling or sputtering sound is heard from the speaker. Set the threshold at the occurrence of spikes. At this point measure the true RMS values of the signal and noise at the input of Noise Adder. Noise performance at J1 = 0 Switch Off the Noise Generator. Set the Volt/div of the scope to 100mv. Slowly vary the potentiometer connected to the FM Input stage such that the Output of it is 400mv Peak – to- Peak. Next connect CH1 to the Output of VCO (Volt/div of the scope = 1v). The displayed waveform is the wide band FM. Now the β ≈ 4. The Carson's bandwidth is 30 khz. Repeat the procedure given in J0 = 0. Find SNR. This SNR is higher than the one calculated at J0 = 0. This clearly brings out the fact that the higher the bandwidth higher the SNR. In FM scheme, one can trade-off between bandwidth and SNR. INFERENCE: The power of FM signal is constant and independent of modulating signal. As modulation index increases the number of significant side band and hence band width increased. RESULT The Generation is Narrow band and Wide band. FM signal was carreid out and FM signal was demodulated. 48/129 49/129 LINE CODING & CARRIER MODULATION TECHNIQUES Aim: 1. To conduct an experiment on Line Coding in DCL – 005 and to observe the various codes broadly as NRZ, Bi phase, URZ. 2. To have two short, comparative, case study on (I) NRZ Vs AMI (II) NRZ Vs Bi phase on the implementation of the above line codes on T1 digital system and to analyses the various transmission related parameters and to suggest proved remedies for the hurdles encountered with supportive graphs. Pre requisites: 1. DCL- 005 (Line coding Kit). 2. CRO. 3. Connecting wires. Theory Principle and purpose behind Line Codes: The transmission of Serial data over any distance immeterial of transmission medium involved, the data has to be maintained since it passes through repeaters, echo chancellors, etc..... The required data integrity must be maintained through data reconstruction with proper timing and retransmission. Line codes were created to facilitate this maintenance. The technique of representing the signal in its electrical form called digital data formats / digital PAM Signals / line codes is called “Line code Techniques”. These line coding techniques can be encoded in several formats. All these PCM Systems can be broadly classified as, 50/129 Unipolar Return to zero format: Non return to zero format: NRZ – L NRZ – M NRZ – S 51/129 A. Return to Zero format ⇒ URZ B. Non – return to Zero formats ⇒ NRZ – L, NRZ-M, NRZ-S C. Phase – encoded format ⇒ Biphasic-L(Manchester), Biphasic-M, Biphasic-S. D. Multi level binary format ⇒ RZ – AMI A. Unipolar Return to zero format: URZ : In URZ One is represented by half – bit wide pulse and zero is represented by absence of pulse. x(t) = A for 0 < t < Tb/2 (half interval) x(t) = 0 for Tb/2 < t < Tb (half interval) B. Non return to zero format: In all the following NRZ formats the transitions takes place at the rising edge of clock pulse. NRZ – L : In NRZ –L One is represented by highs and zeros by lows x (t) = 1 for O < t < Tb (full interval) x (t) = O for O < t < Tb (full interval) NRZ – M : One is represented by change in levels and zeros by no transition. NRZ – S : One is represented by no transitions and zero is represented by change in levels. C. Phase Encoded formats: The phase Encoded signals are special in the sense that they are composed of both in phase and out of phase component of the clock. In all the formats transition occurs at the beginning of every bit interval. If ‘1’ is transmitted x (t) = A/2 for O < t < Tb/2 x(t) =-A/2 for Tb/2 < t < Tb 52/129 Phase Encoded formats: Biphase - L: Biphase – M: Biphase – S: Multilevel Signals: RZ – Alternate mark inversion (RZ – AMI) 53/129 If ‘0’ is transmitted x (t) = -A/2 for O < t < Tb/2 x(t) = A/2 for Tb/2 < t < Tb. Biphase - L : One is represented by half bit wide pulse positioned during first half of the bit interval and a zero represented by half bit wide pulse positioned during second half of the bit interval. Biphase – M: One is represented as Second transition at half bit later whereas zero has no Second transition. Biphase – S: One is represented by no second transition. Zero is represented by Second transition a half bit later. D. Multilevel Signals : Multi level signals use one or more levels of voltages to represent the binary digits ones and zeros. The commonly used multi – level signal is AMI. RZ – Alternate mark inversion (RZ – AMI ) One is represented by equal amplitude of alternate pulses and zero by no pulses. General Inference on various coding formats: 1. The advantage of RZ – AMI is that the ambiguities due to transmission sign inversion are eliminated. 2. In all bipolar cases it needs absolute sense of polarity at the receiver. 4. It is obvious from the waveforms observed that NRZ – L,M,S have the frequency component equal to half the clock frequency. 5. Whereas Biphase – L,M,S, URZ and AMI have the frequency of equal to that of clock . 1. URZ & AMI signals are composed of both in phase and out of phase components of the clock. 54/129 55/129 THE ABOVE FORMATS COULD BE COMPARED AND ANALYSED BASED ON THE VARIOUS TRANSMISSION PARAMETERS AS FOLLOWINGLY. Parameters Unipolar RZ Bipolar NRZ (AMI) Bi phase - L (Manchester) Polarity 0, +ve two levels 0, +ve, -ve three levels +ve, -ve two levels Availability of D.C component present absent absent Signal freg Bandwidth fb (higher Bandwidth) fb/2 fb (higher Bandwidth) Peak power requirement high high low Noise Immunity poor good good Cross talk high low low Synchronization Capability (effect of string of 1’S / 0’S) better good good Timing no timing information no timing information good clock recovery Experimental Procedure : The DCL – 005 trainer initiates the various data conditioning techniques normally adopted in practice. The system broadly illustrates the following blocks. (i) Data Simulator Logic (ii) Coding logic 56/129 57/129 Data Simulator Logic : The data simulator logic generates NRZ – L data a constant pattern and a reference clock (250 KHZ) Which is the origin for all the other data formats. Coding Logic : The coding Logic converts NRZ – L data to NRZ – M, NRZ – S, URZ, Biphase – L, Biphase – M, Biphase – S and RZ – AMI Signals. Set DCL – 005 in stand alone mode power on the unit and connect the internally generated clock and data to the relevant sockets of the data coding modules. Observation : Observe the various waveforms anticipated at the various test points provided. Observe the NRZ-L data and the various NRZ-M, NRZ-S, URZ, Biphase – L, Biphase – M, Biphase – S, AMI. When a larger no of such PCM signals has to be transmitted over a common channel, multiplexing of PCM signals is required.Let us have a case study of multiplexing PCM signals Bipolar NRZ & AMI over T1 Digital System. STUDY NO:1 Case study of multiplexing PCM signals Bipolar NRZ & AMI over T1 Digital System. Specifications of T1 digital system : It is a Simple transmission over telephone line using wide band Co-axial Cable. � It accommodates 24 analog signals from S1 – S24. � Each signal is band limited to approximately 3.3 KHZ and is sampled at the rate of 8 KHZ, the Nyquist rate of 2 x 3.3 = 6.6 KHZ. � The digital signal generated during the course of one complete sweep is therefore 24x8 = 192 bits. 58/129 Fig 1.8 59/129 � Synchronization is achieved by adding a single bit preceding the 192 bits. So that to form a frame of 193 bits. � Bit rate : Tp = 1/8000 = 125 Micro secs. Bit rate of T1 System : 193 M bits/s = 1.544 M bits/s 125 The Inferences of NRZ wave form over T1 Digital System For the variety of practical reasons the low frequency response of T1 channel is limited hence we simulate a rudimentary low frequency RC network for the transformer, series capacitances etc., through which signal is supposed to pass. The output response is as shown � When the input is constant the output decays exponentially (because of RC time constant) to zero volts. � When the input changes abruptly by a amount of 2v the output changes by an equal amount since instantaneous change in Capacitor is not possible. At the receiver end : The most effective way of distinguishing at the receiver, ‘1’ from ‘0’ is measuring the area under the waveform. A positive area indicates I and negative area as ‘0’ Long persistency of “1” causes Severe depression When the sequences has a long 1’s the equivalent area available is severely depressed. The same complicacy occurs in case of long O`S also.If there is long stream of ones/zeros.The receiver could conceivably suffer so much compound jitter that is would either loss or gain an entire bit time, and then he out of sync with the Transmitter as shown in Fig:1.8 Result of the study: Hence in a NRZ waveform the long persistence sequence of1’s / 0’s builds up a “D.C component” which cannot be transmitted over a T1 channel. 60/129 Fig no 2. 61/129 The remedy is RZ-AMI of Multi level binary format where the 1’S will be represented by the bit of half pulse width with alternate polarity and 0’S by no transition hence neither the persistence sequence of 1`s nor 0`s will generate the accumulation of D.C component” as shown in Fig:2.If the receiver should encounter two successive pulses of same polarity then it will be considered as the violation of waveform standards. The Inference of AMI in T1 Digital system : In AMI to prevent the bit Synchronizing from drop out of synchronism and to guarantee an accurate operation it is necessary that there may not be extended periods of waveforms w/o any transition (in case of long string of O’S) which shall affect the precision of synchronization the following remedy could be implemented. Remedy for the Synchronization problem in RZ-AMI : A procedure which circumvents this difficulty involve delaying the AMI waveforms briefly before transmission. So that it can be monitored. Whenever a long string of zero’s is encountered the string of zeros will be replaced by a coded waveform containing frequent 1’S.The 1’S are inserted without any alternates which shall enable us to distinguish from the original 1’s to keep the Synchronize going. STUDY NO:2 COMPARATIVE STUDY ON NRZ Vs BI - PHASE : In NRZ as shown in the sequence the continues 1’S are represented by a sustained fixed level of +Vb or –Vb voltage never returns to zero. 62/129 Power Spectral densities of NRZ Power Spectral density in case of Biphase 63/129 In case of Biphase the signal is identical to clock waveform or reversed from the clock depending on whether 1’S or 0’S.Hence the signal passes through zero at least once per clock cycle. Since the Biphase changes its level more frequently than NRZ the Biphase spectrum naturally have more high frequency component. Let us compare the power Spectral densities of NRZ and Biphase Power Spectral densities of NRZ G(f) = IP(f)I2 Ts G (f) = X2b Tb 2 fTb Sin fTb π π 1. It has a maximum at f = 0, and G (f) = 0 at all multiples of (fb = 1/Tb) The pedk occurs between fb and 2fb ~ 1.5 fb and 14 dB lower than the peak at f = 0 hence the total power in an MRZ signal is reduced only by 10% if passed through LPF with the cutoff frequency f = fb. Thus 90% of power is at main lobe centered at f = 0 is saved moreover it doesn’t contains any D.C. Component. 2. Power Spectral density in case of Biphase = G(f) = V2b Tb 2 2 / 2 / 2 fTb Sin fTb π π Inference : The principal lobes extends from (f = 0, to f = 2fb) and has the peaks approximately ~ 3fb/4 and at f = 0, G(f) = 0 It can be verified that if Bi phase Signal is transmitted through a LPF with a cut off frequency at 2fb the more than 95% of power is passed and results in safe transmission. Again we have to compromise that the cost in case of Bi phase since the modulation rate is #twice that of NRZ. The possible solution could be Quaternary coding where two successive bits are coupled together to get represented by a single symbol / Voltage level. The analysing and pros & cons of the same is forced to get restricted because of scope of the objective. 64/129 Clock with NRZ-L Wave Form 65/129 Conclusion for both the studies : The line codes have various advantages like 1. Robustness to channel noise and interference 2. Efficient regeneration of coded signal along the transmission path. 3. Efficient exchange of increased channel bandwidth for input S/N ratio by obeying exponential law. 4 Secure communication through the use of encryption / decryption modulation schemes. 5. Error detection capability and matched power spectrum. 5. Transparency of availability is & O’S over the channel probable drawbacks and proved remedies. 6. All the above advantage are attained at the cost of increased system complicacy and Increased Channel Bandwidth. (i) Though many complicacies are faced in the design, comerically available VLSI chips stands granted. (ii) Turning to the issue of bandwidth we do recognize that the increased bandwidth requirement may be reason for justifiable concern in past. The increased availability of wide band communication channels means that bandwidth is no more a system constraint Thanks for the deployment of communication satellites for broad casting and fibre optic communications for liberating the bandwidth constraint. (iii) Though with the Sophisticated data compression techniques it is indeed possible to remove the redundancy inherently present they reduce the bit rate of transmission without serious degradation in System performance. 66/129 NRZ-L and BIO-S WaveForm 67/129 Eventually to say on broader angle the action of recuperating the clock signal from the received signal is inevitably affected by some deterioration. The square wave extracted is not exactly synchronous with the transmitted clock. Timing information, that is essentially carried by the level transitions of the received signal has been affected by the noise and the ISI and has acquired, i) Some inevitable delay, due to physical transit time. ii) Some inaccuracy = Phase moudulation, called jitters, which could be only minimised but not eliminated. 9. bit error rate of very low value < 10-19 Nevretheless, comparing to the complication involved in analog transmission and up-dation in the circuitaries, though we compromise in the originality of the signal in digital as we know that digital has its own advantages, flexibilities and the power of Digital signal processing etc..... Result: Thus the experiment and the relevant case studies has been completed successfully. Carrier Generation Logic: The carrier waves are generated in synchronization with the reference clock to produce the three carrier sine waves that are necessary for carrier modulation. A PLL is used to synchronize the carrier with the incoming clock. The modulation logic modulates the carrier with the digital information signals. The various blocks that can be identified in DCl-006 are: Detection Logic Data Decoding Logic 68/129 NRZ-L With ASK Modulation 69/129 1. Detection Logic a). ASK Demodulator: This detector is built around a diode rectifier and a threshold detector. Whenever the sine wave is transmitted, the detector identifies one and whenever the carrier is rejected from transmission, the detector identifies it as a zero. b). FSK Demodulator: The digital PLL forms the heart of this logic. The PLL center frequency and the lock range are fixed around 2 MHz. So whenever the 2 MHz signal is transmitted, then the phase detector output of the PLL remains low. Thus the phase detector output of the PLL directly gives the demodulated data. c) PSK Demodulator: The first step in PSK demodulation is the sine to square wave conversion using a Schmitt trigger. The PSK demodulator works on the principle of square law. The digital PLL IC is used for doubling the frequency component of the data. A phase comparator logic built around flip flops demodulates the data modulated carrier. EXPERIMENTAL PROCEDURE: ASK Modulation: Experimental Setup: Setup DCL-005 in stand alone Connect S-Clock to the Coding Clock Connect S-Data to the Control Input of the Modulator Connect one channel of the scope to the S-Data. If it is not matching with the S- Data slightly adjust the potentiometer P! Connect Sin 1 to the Input 1 of the modulator. The amplitude of the sine wave can be varied by means of potentiometer P$. Connect GND to Input 2 of the modulator Connect the Scope to the Control Unit test point and observe the modulated output with respect to control input. Inference: We observe that the ASK output resembles the AM signals waveforms. Only NRZ-L is used. ASK Demodulation: Experimental Setup: Maintain the connections in the DCL-005 as in the previous experiment Connect the ASK output to the ASK demodulator input in the DCL-006 70/129 NRZ-L With FSK Modulation 71/129 Connect the Scope to the Ask input and the other channel to the data output of the ASK demodulator Observations: Observe the ASK modulated output and the modulating data in the two channels of the oscilloscope. Also observe the modulated carrier at the input of DCL-006. Observe the ASK demodulated output at the Data Output of the ASK demodulator and compare the demodulated signal with respect to the control input. Inference: We observe that a very small time lag between the modulating data and the recovered data which is less than one half of the carrier time period. FSK Modulation: Experimental Setup: Connect the Sin 1 to the Input 1 of the Modulator Connect Sin 2 to the Input 2 of the Modulator Connect data to the Control Input of the Modulator Connect the scope to the Control Input and the other channel to the Modulated output. Observations: Observe the FSK Modulated output and the modulating data in the two channels of the oscilloscope. Observe the incoming modulated carrier and the recovered data with respect to the modulating data. Inference: Since the tracking ability and the time response of the PLL is limited, a small phase lag exists between the recovered data and the modulating data. FSK Demodulation: Experimental Setup: Establish the same connections for FSK modulation in DCL 005 Set DCL 005 in conjunction with DCL 006 Connect the FSK modulated output to the FSL input of the FSK demodulator (DCL 006) Connect the scope to the control input and the data output 72/129 NRZ-L with PSK Modulation 73/129 Observations: Observe the FSK modulated output and the modulating data in the two channels of the scope. Observe the incoming modulated carrier and the recovered data with respect to the modulating data. Inference: Since the tracking ability and the time response of the PLL is limited, a small phase lag exists between the recovered data and the modulating data. PSK Modulation: In the PSK modulation for all one to zero transitions of the modulating data, the modulated output switches between the in phase and out of phase components of the modulating frequency. The frequency and phase components chosen for BPSK modulation are as follows. 4. 0.5 MHz (0 degrees) sine wave carrier for representing 1 5. 1 MHz (180 degrees) sine wave carrier for representing 0 Experimental Setup: Maintain the setup as for the other keying equipments Connect the Sin2 to Input 1 of the modulator. The modulator of Sin 2 can be adjusted by means of the Pot P2 Connect the Sin2* to Input 2 of the modulator. The modulator of Sin 2 can be adjusted by means of the Pot P3 Connect the scope to the control input of the modulator and the modulated output Observations: Observe the PSK Modulated output with respect to the control input. Observe the phase shifts in the frequency during each transition in the data PSK Demodulation: Experimental Setup: Establish the same connections for PSK modulation and connect the DCL 006 in conjunction with the DCL 005. Feed the PSK modulated output to the PSK input of the PSK demodulator of the DCL 006 Connect the scope to the PSK input and the data output for PSK demodulated output Observe the PSK demodulated output with respect to the control input of the modulator 74/129 75/129 Observations: Observe the PSK modulated output and the modulating data in the two channels of the scope. Observe the incoming modulated carrier. Observe the extracted reference carrier with reference to the in phase modulating carrier. Also observe the recovered data with respect to modulating data. Inference: It is observed that the successful operation of the PSK is fully dependent on the phase components of the transmitted modulated carrier. If the phase reversals of the modulated carrier along with the rising edges and falling edges of the data are not proper, then the efficient detection of data from PSK modulated carrier becomes impossible. RESULT: The different line coding techniques were studied and understood. The various Carrier Modulation and Demodulation schemes involved in the digital communication system was studied. 76/129 % Amplitude Modulation MATLAB Code % clear all;close all; Fs = 1/.0001; % Sampling rate t=0.0001:0.0001:1; Nt = length(t); % Number of sample points Nf = 2^18; % Number of points in FFT, for faster computation f = (0:Nf-1)'/Nf*Fs; % Frequencies in FFT % Construct message signal m=1*sin(2*pi*10*t);M = fft(m, Nf); figure(1);subplot(211);plot(t,m) title('Message signal m(t)');xlabel('Time (sec)') subplot(212);plot(f, abs(M)); ax = axis;axis([0 Fs/400 ax(3) ax(4)]); % Plot up to one-half the sampling rate title('Message spectrum M(f)');xlabel('Frequency (Hz)'); % Define carrier frequency and amplitude for modulation fc = 1000;Ac = 1;c = Ac * cos(2*pi*fc*t); % Carrier wave, unmodulated ka = 1.5; % modulation index sAM = (1 + ka*m) .* c;SAM = fft(sAM, Nf); figure(2);subplot(221);plot(t,sAM); title('AM modulated signal');xlabel('Time (sec)') subplot(223);plot(f, abs(SAM)); ax = axis;axis([0 Fs/5 ax(3) ax(4)]); title('Spectrum of AM signal');xlabel('Frequency (Hz)') % Envelope detector for AM sAMplus = hilbert(sAM); % Pre-envelope = s(t) + j * \hat{s}(t) mAMdemod = sAMplus .* exp(-j*2*pi*fc*t); % Complex envelope % Envelope detector output MAMdemod = fft(mAMdemod, Nf); subplot(222);plot(t,abs(mAMdemod)) title('Envelope detector output');xlabel('Time (sec)') subplot(224);plot(f, abs(MAMdemod)) ax = axis;axis([0 Fs/400 ax(3) ax(4)]); title('Spectrum of envelope detector output');xlabel('Frequency (Hz)') % DSB-SC modulation sDSB = m .* c; SDSB = fft(sDSB, Nf); figure(3);subplot(221);plot(t,sDSB) title('DSB-SC modulated signal');xlabel('Time (sec)') subplot(223);plot(f, abs(SDSB)); ax = axis;axis([0 Fs/5 ax(3) ax(4)]); title('Spectrum of DSB-SC signal');xlabel('Frequency (Hz)') % Coherent demodulator phi = pi/6; % Phase of local oscillator local = cos(2*pi*fc*t + phi); vDSB = sDSB .* local; Fcut = 1000;order = 5; Fdig = Fcut / (Fs/2); % "Digital frequency" normalized by sampling rate [b,a] = butter(order, Fdig); mDSBdemod = filter(b, a, vDSB); % Apply the filter to the vDSB signal mDSBdemod = mDSBdemod - mean(mDSBdemod); MDSBdemod = fft(mDSBdemod, Nf); subplot(222);plot(t,mDSBdemod); title('Coherent detector output');xlabel('Time (sec)') subplot(224);plot(f, abs(MDSBdemod)); ax = axis;axis([0 Fs/400 ax(3) ax(4)]); title('Spectrum of coherent detector output');xlabel('Frequency (Hz)') 77/129 Generate AM and FM modulated waveform using MATLAB Aim: To Generate the AM and FM modulated waveforms and analyzing those modulation performance using the MATLAB 7.0 software utility. Apparatus Required: 1. PC with MATLAB SOFTWARE. Theory: Amplitude Modulation In amplitude Modulation amplitude of the sinusoidal carrier wave is varied in accordance with the base band signal. The following figure shows the sinusoidal amplitude modulated waveforms. Figure(a) shows the carrier waveform, figure(b) shows the modulating waveform and c shows the modulated waveform. Consider a sinusoidal carrier wave c(t) is defined by c(t) = Ac.Cos(2πfct) 78/129 Message Signal with Frequency Spectrum 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -1 -0.5 0 0.5 1 Message signal m(t) Time (sec) 0 5 10 15 20 25 0 1000 2000 3000 4000 5000 6000 Message spectrum M(f) Frequency (Hz) AM Under Modulation Ka=0.5 0 0.2 0.4 0.6 0.8 1 -1.5 -1 -0.5 0 0.5 1 1.5 AM modulated signal Time (sec) 980 990 1000 1010 1020 0 1000 2000 3000 4000 5000 Spectrum of AM signal Frequency (Hz) 0 0.2 0.4 0.6 0.8 1 0.5 1 1.5 Envelope detector output Time (sec) 0 5 10 15 20 25 0 2000 4000 6000 8000 10000 Spectrum of envelope detector output Frequency (Hz) 79/129 where Ac is the carrier amplitude and fc is the carrier frequency. In that the above equation phase of the carrier wave is zero. Let m(t) denotes the base band signal that carries the specification of the message. An amplitude modulation (AM) is defined as a process in which the amplitude of the carrier wave c(t) is varied about a mean value, linearly with the base band signal m(t). an amplitude modulated (AM) wave may thus be described in its most general form as a function of time as follows: s(t) = Ac* [ (1+ka*M(t) * cos(2πfct) ] where ka is the constant called the amplitude sensitivity of the modulator responsible for the generation of the modulated signal s(t). The following figure shows a base band signal m(t), and corresponding AM wave s(t) for two values of amplitude sensitivity ka(modulation index) and a carrier amplitude Ac=1 volt. We observe that the envelope of s(t) has essentially the same shape as the base band signal m(t) provided that two requirements are satisfied: � The amplitude of kam(t) is always less than unity, that is, kam(t) < 1 for all t. This condition is illustrated in figure. It ensures that the function 1+kam(t) is always positive, and since an envelope is a positive function. We may express the envelope of the AM wave s(t) as Ac(1+kam(t)).This is called under modulation of the AM. � When the amplitude sensitivity of the ka of the modulator is large enough to make � kam(t) >1 for any t is called as over modulated, resulting the carrier phase reversal occurs wherever the m(t) crosses the zero. � The carrier frequency fc is very larger than the maximum frequency component in the message signal. fc>>W if the condition is not satisfied an envelope cannot be visualized satisfactorily. 80/129 0 0.2 0.4 0.6 0.8 1 -2 -1 0 1 2 AM modulated signal Time (sec) 980 990 1000 1010 1020 0 1000 2000 3000 4000 5000 Spectrum of AM signal Frequency (Hz) 0 0.2 0.4 0.6 0.8 1 0 0.5 1 1.5 2 Envelope detector output Time (sec) 0 5 10 15 20 25 0 2000 4000 6000 8000 10000 Spectrum of envelope detector output Frequency (Hz) AM Perfect Modulation Ka=1.0 AM Over Modulation Ka=1.5 0 0.2 0.4 0.6 0.8 1 -3 -2 -1 0 1 2 3 AM modulated signal Time (sec) 980 990 1000 1010 1020 0 1000 2000 3000 4000 5000 Spectrum of AM signal Frequency (Hz) 0 0.2 0.4 0.6 0.8 1 0 0.5 1 1.5 2 2.5 3 Envelope detector output Time (sec) 0 5 10 15 20 25 0 2000 4000 6000 8000 10000 Spectrum of envelope detector output Frequency (Hz) 81/129 The Fourier transform of AM wave s(t) is given by S(f) = Ac/2 [d(f-fc) + d(f-fc)] + kaAc/2 [M(f-fc) + M(f +fc)] This spectrum consists of two delta functions weighted by the factor Ac/2 and occurring at ± fc ; and two versions of the baseband spectrum translated in frequency by ± fc and scaled in amplitude by kaAc/2 . 82/129 AM- DSB –SC Signal and Coherent Detector output With Frequency spectrum 0 0.2 0.4 0.6 0.8 1 -1 -0.5 0 0.5 1 DSB-SC modulated signal Time (sec) 980 990 1000 1010 1020 0 500 1000 1500 2000 2500 3000 Spectrum of DSB-SC signal Frequency (Hz) 0 0.2 0.4 0.6 0.8 1 -0.5 0 0.5 Coherent detector output Time (sec) 0 5 10 15 20 25 0 500 1000 1500 2000 2500 Spectrum of coherent detector output Frequency (Hz) 83/129 From the spectrum of figure we note the following points: � As a result of modulation process , the spectrum of the message signal m(t) for completely including negative sides in message signal also becomes visible for positive frequencies, provided the condition fc>>W. � In the positive frequencies of the spectrum having the two side bands called upper side band and lower side band. � The band width of the AM wave which exactly twice the message band width W, that is Bt = 2W. Demodulations: The demodulation is the process of detecting the original signal from the received signal. This received signal consists of noise & original signal. There are three types of received format 1.Envelope Detector. 2.Squarelaw Detector. Limitations: � Amplitude modulation is wasteful of power. � This is modulation occupies the more bandwidth. Twice the message bandwidth. Frequency Modulation The frequency modulation is that form of angle modulation in which the instantaneous frequency fi(t) is varied linearly with the message signal m(t) , as shown by fi(t) = fc + kf*m(t). The term fc represents the frequency of the unmodulated carrier, and the constant kf represents the frequency sensitivity of the modulator expressed in Hertz per volt. The FM modulation is the nonlinear modulation process so that the spectrum of the FM modulation is not simple manner so that analysis of FM is difficult process comparative AM modulation. Here we consider two types of FM called namely 2. Narrow band FM. 3. Wide band FM. 84/129 % Frequency Modulation MATLAB Code % close all;clear all;clc dt = 1e-4; % Sample spacing (seconds) Fs = 1/dt; % Sampling rate t = (dt:dt:2)'; % Sample times: total of 2 seconds of data Nt = length(t); % Number of sample points Nf = 2^18; % Number of points in FFT, for faster computation f = (0:Nf-1)'/Nf*Fs; % Frequencies in FFT % Construct message signal Am=1;mf=10;m=Am*cos(2*pi*mf*t);M = fft(m, Nf); figure(1);subplot(211);plot(t,m); title('Message signal m(t)');xlabel('Time (sec)'); subplot(212);plot(f, abs(M)); ax = axis; axis([0 Fs/400 ax(3) ax(4)]);% Plot up to one-half the sampling rate title('Message spectrum M(f)');xlabel('Frequency (Hz)') % Define carrier frequency and amplitude for modulation fc = 1000;Ac = 1;c = Ac * cos(2*pi*fc*t); % kf = 1.5; % Frequency sensitivity int_m(1) = 0; % Integrate the signal to produce FM signal for k=1:Nt-1 int_m(k+1) = int_m(k) + m(k)*dt; end sFM = Ac * cos(2*pi*fc*t + 2*pi*kf*100*int_m'); SFM = fft(sFM, Nf); figure(2);subplot(221);plot(t,sFM); title('FM modulated signal');xlabel('Time (sec)'); subplot(223);plot(f, abs(SFM)); ax = axis;axis([0 Fs/5 ax(3) ax(4)]); title('Spectrum of FM signal');xlabel('Frequency (Hz)') % FM demodulator: Derivative followed by envelope detector dsFM = sFM(1); % Compute derivative for k=2:Nt dsFM(k) = (sFM(k)-sFM(k-1))/dt; end % Envelope detector dsFMplus = hilbert(dsFM); % Pre-envelope dsFMtilde = dsFMplus .* exp(-j*2*pi*fc*(t')); % Complex envelope mFMdemod = abs(dsFMtilde); % Envelope detector output mFMdemod = mFMdemod - mean(mFMdemod); % Remove DC from output MFMdemod = fft(mFMdemod, Nf); subplot(222);plot(t,mFMdemod); axis([0 2 kf*-600 kf*600]); title('FM detector output');xlabel('Time (sec)'); subplot(224);plot(f, abs(MFMdemod)); ax = axis;axis([0 Fs/400 ax(3) ax(4)]); title('Spectrum of FM detector output');xlabel('Frequency (Hz)'); 85/129 Consider then a sinusoidal modulating signal defined by m(t) = Am cos(2*pi*fm*t) The instantaneous frequency of the resulting FM signal equals fi(t) = fc + kf*Am*cos(2*pi*fm*t) fi(t) = fc + Δf cos(2π*fm*t) where Δf = kf*Am The quantity Δf is called the frequency deviation , representing the maximum departure of the instantaneous frequency of the FM signal from the carrier frequency fc. A fundamental characteristic of an FM signal is that the frequency deviation Δf is proportional to the amplitude of the modulating frequency. The fm signal is obtained as θi(t) = 2πfct + Δf/fm sin(2πfmt). The ratio of the frequency deviation Δf to the modulation frequency fm is commonly called the modulation index of the FM signal. We denote it by the letter β and so write β = Δf/fm The parameter β represents the phase deviation of the FM signal, that is the maximum departure of the angle θi(t) from the angle 2πfct of the unmodulated carrier, hence β is measured in radians The FM signal itself is given by s(t) = Ac cos[2πfct + βsin(2πfmt)] Depending upon the value of β we differentiate the Narrow band and Wide band FM. � If β is small compared to one radian for Narrow band FM � If β is large compared to one radian for wide band FM 86/129 Message Signal with Frequency Spectrum 0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 -1 -0.5 0 0.5 1 Message signal m(t) Time (sec) 0 5 10 15 20 25 0 2000 4000 6000 8000 10000 Message spectrum M(f) Frequency (Hz) Narrow band FM Kf=0.1 0 0.5 1 1.5 2 -1 -0.5 0 0.5 1 FM modulated signal Time (sec) 980 990 1000 1010 1020 0 2000 4000 6000 8000 Spectrum of FM signal Frequency (Hz) 0 0.5 1 1.5 2 -60 -40 -20 0 20 40 60 FM detector output Time (sec) 0 5 10 15 20 25 0 1 2 3 4 5 6 x 105 Spectrum of FM detector output Frequency (Hz) 87/129 Spectrum of wide band FM: The Fourier transform of the both FM is S(f) = Ac/2 Σ Jn(β)[d(f-fc-nfm) + d(f+fc+nfm)] in that equation Jn(β) is the Bessel function. This Bessel function gives the characteristics of the FM modulated waveform properties. The bandwidth of the FM modulation is decided by carsons rule equation (or) universal graph. If modulation index increase the side bandwidth of the FM bandwidth also increased. B = 2 Δf +2fm = 2 Δf(1+1/ β). 88/129 Wide Band FM Kf=1.5 0 0.5 1 1.5 2 -1 -0.5 0 0.5 1 FM modulated signal Time (sec) 600 700 800 900 1000 1100 1200 1300 0 500 1000 1500 2000 2500 3000 Spectrum of FM signal Frequency (Hz) 0 0.5 1 1.5 2 -500 0 500 FM detector output Time (sec) 0 5 10 15 20 25 0 2 4 6 8 10 x 106 Spectrum of FM detector output Frequency (Hz) 89/129 Generation of wideband FM: A simplified block diagram of an indirect FM system is shown in the figure. The message signal m(t) is first integrated and then used to phase modulation a crystal controlled oscillator ; the use of crystal oscillator provides the frequency stability. To minimize the distortion inherent in th phase modulator, a maximum phase deviation or modulation index B is kept small. From this we will generate the wide band FM by means of a frequency multiplier so as to produce the desired wide band FM signal. The modulated wave form S(t) = Ac cos[2πfct + 2πkf (integration(m(t)dt)]. FM demodulation: Frequency demodulation is the process that enables us to recover the original modulating signal from a frequency-modulated signal. The objective is to produce a transfer characteristic that is the inverse of that of the frequency modulator, which can be realized directly or indirectly. The above diagram shows the direct method of frequency demodulation involving the popular device known as frequency discriminator, whose instantaneous output amplitude is directly proportional to the instantaneous frequency of the input FM signal. The above figure shows the ideal frequency discriminator as a pair of slope circuits with their complex transfer functions related by the equation H2 (f) = H1 (-f) Followed by envelope detectors and finally a summer,. This scheme is called a balance frequency discriminator. Algorithm for Generating AM modulated Wave: 1. Getting the inputs like Modulation index, Modulating frequency, Carrier frequency, & amplitude of the carrier and modulating signal. 2. Generate the modulating waveform as per input specifications. 3. Generate the carrier waveform as per input specifications. 4. Plot the both Modulating and carrier wave form in the same figure using the subplot command. 5. Generate the modulated waveform by using the following equation S(t) = Ac * (1+ MI*m(t)) * cos (2 *pi*fc*t). 6. Plot the modulated waveform. 90/129 Generation of wideband FM: FM demodulation: 91/129 Algorithm for Generating AM Demodulated Wave: 7. Taking the Hilbert transform for the modulated waveform. 8. The demodulated output was taken by using the following equation. Demod = (HilberTransform(s(t))) * (exp(-j*2*pi*CF*t)). 9. Plot the demodulated output in separate figure. Algorithm for Generating FM modulated Wave: 1. Getting the inputs like frequency sensitivity, Modulating frequency, Carrier frequency, & amplitude of the carrier and modulating signal. 2. Generate the modulating waveform as per input specifications. 3. Generate the carrier waveform as per input specifications. 4. Plot the both Modulating and carrier wave form in the same figure using the subplot command. 5. Generate the modulated waveform after integrating the modulating signal and then multiplying with the carrier. 6. Plot the modulated waveform in the figure. Algorithm for Generating FM Demodulated Wave: 7. Taking the Hilbert transform for the modulated waveform the demodulation is started. 8. The demodulated output is taken by using an envelope detector. 9. The demodulated output is plotted in a separate figure. 10. The Spectral content of the modulated waveform is also plotted. Result: Thus the modulation and demodulation of amplitude modulation, frequency modulation using MATLAB was performed. The spectral content of the modulated waveform is also studied 92/129 Generation of Coherent Binary FSK Signals: Detection of Coherent Binary FSK Signals: 93/129 Frequency Shift Keying (FSK) Using Mat Lab Software Simulation Aim: The Frequency shift keying studying and stimulation using Mat Lab Software and plot the Modulated, demodulated outputs and performance. Requisites: The Mat Lab Software installed Computer Theory: In a coherent binary FSK system, the pair of signals S1(t) and S2(t) used to represent binary symbols 1 and 0, respectively, is defined by S1(t) = √2 Eb / Tb cos(2πf1t) (1) S2(t) = √2 Eb / Tb cos(2π(f1+f)t = √2 Eb / Tb cos(2π(f2)t (2) where 0 ≤ t < Tb, and Eb is the transmitted signal energy per bit. The typical pair of sinusoidal waves are described by ..(3) Where i=1,2 and transmitted frequency fi is From the equation (1) and (2) ,it is clear that S1(t) and S2(t) is orthogonal, but not normalized to have unit energy. ..(4) Where i=1,2 94/129 Frequency Shift Keying Code for Modulation and Demodulation %Initilization: numbits=input('Enter the number of data samples:'); f1=200; % in hz f=1000; % in hz stime = 0.005; mid= 0.0001; t=0:mid:stime-mid; ipseq=randint(1,numbits); out=[]; %Modulation: for i = 1 : length(ipseq) out=[out sin(2*pi*(f1+ipseq(i)*f)*t)]; end subplot(3,1,1); stem(ipseq); title('input sequence'); subplot(3,1,2); plot(out); title('output fsk'); %Demodulation: demod=[]; a=sin(2*pi*rootf*t); for i = 1:length(ipseq) y=(i-1)*50; d=asin(out(y+1:y+50)); 95/129 Correspondingly, the co-efficient Sij for i=1,2 and j=1,2 is defined by ..(5) Two message points is defined as And ..(6) With the Euclidean distance between them equal to √2Eb . Signal space diagram for FSK: Generation of Coherent Binary FSK Signals: To generate the binary FSK input binary data sequence is first applied to ON/OFF level encoder, the output of the symbol 1 is √E b , and output of symbol 0 is 0. This signal is forwarded to both upper channel and lower channel. By using inverter in lower channel, any one channel will switch closed other will open 96/129 d1=d(2)/a(2); if d1 < 2 demod = [demod 0]; else demod=[demod 1]; end end subplot(3,1,3); stem(demod); title('demodulated output'); %FFT: %It is difficult to identify the frequency components by looking at the original signal. %Converting to the frequency domain, the discrete Fourier transform of the noisy signal %found by taking the 1024-point fast Fourier transform (FFT): Y = fft(out,1024); %The power spectrum, a measurement of the power at various frequencies, is Pyy = Y.* conj(Y) / 1024; %Graph the first 257 points (the other 255 points are redundant) on a meaningful frequency axis: f = 8400*(0:512)/1024; figure(2);%subplot(4,1,3); plot(f,Pyy(1:513)); title('Frequency content of modulated signal'); xlabel('frequency (Hz)'); 97/129 ie. For symbol 1: Upper channel closed, √E b multiplied with Φ1(t) ,whereas lower channel is open. Result is √2 Eb / Tb cos(2πf1t) is transmitted. ie. For symbol 0: Lower channel closed, √E b multiplied with Φ2(t) ,whereas upper channel is open. result is √2 Eb / Tb cos(2πf2t) is transmitted. Detection of Coherent Binary FSK Signals: To detect the transmitted binary sequence ,input signal x(t) is product with Φ1(t) in upper channel and integrated and x(t) is product with Φ2(t) in lower channel and integrated .Resultant of both channel is forwarded to decision making device. Where y > 0= Symbol 1 y < 0 = Symbol 0 Algorithm Transmitter or Modulation 1. Generate the no of Random sequence bits as input given 2. Initialize frequency f1 and f 3. Initialize time limit for integration and other parameters 4. To generate the modulated output f1 or f2 multiply input bit sequence with f. 5. If input sequence is Eb, f will present ,otherwise it will be zero. 6. Add f1 and f to generate transmitted signal i.e f1= f1+0 (where f = 0) f2 = f1+f 7. Plot the input data and modulated carrier Receiver or Demodulation 1. The Received signal is multiply with the same basis function. 2. sum the multiplied signal at the Tb interval (i.e. integration ) 3. Compare the summation data for > 2 or < 2. If >2 make Symbol 1 and < 2 Make Symbol 0. 98/129 0 500 1000 1500 2000 2500 3000 3500 4000 4500 0 5 10 15 20 25 30 35 40 45 50 Frequency content of modulated signal frequency (Hz) Plot for input sequence, Modulated and Demodulated 0 2 4 6 8 10 12 14 16 0 0.2 0.4 0.6 0.8 1 input sequence 0 100 200 300 400 500 600 700 800 -1 -0.5 0 0.5 1 output fsk 0 2 4 6 8 10 12 14 16 0 0.5 1 demodulated output Plot for FFT: 99/129 4. Calculate FFT for signal 5. Plot the received bit and FFT. Inference • It has two basis function Φ1(t) and Φ2(t) . • S1(t) and S2(t) is orthogonal, but not normalized to have unit energy. • Transmitted signal frequency contain only f1 (f1=f1+0) or f2 (f2= f1+f) • In demodulation, Input signal is product with two basis function and integrated to Detect symbol 1or 0. Result The Frequency shift keying Transmitter and Receiver understood and analyzed and plotted the FFT performance. 100/129 Block diagrams for (a) binary PSK transmitter and (b) coherent binary PSK receiver 101/129 Binary Phase Shift Keying (BPSK) Using Mat Lab Software Simulation Aim: The Binary Phase shift keying studying and stimulation using Mat Lab Software and plot the Modulated, demodulated outputs and performance. Requisites: The Mat Lab Software installed Computer Theory: In a coherent binary PSK system, the pair of signals S1(t) and S2(t) used to represent binary symbols 1 and 0, respectively, is defined by S1(t) = √2 Eb / Tb cos(2πfct) (1) S2(t) = √2 Eb / Tb cos(2πfct + π) = - √2 Eb / Tb cos(2πfct) (2) where 0 ≤ t < Tb, and Eb is the transmitted signal energy per bit. To ensure that each transmitted bit contains an integral number of cycles of the carrier wave, the carrier frequency fc is chosen equal to nc / Tb for some fixed integer nc. A pair of sinusoidal waves that differ only in a relative phase-shift of 180 degrees, as defined in Equations (1) and (2), are referred to as antipodal signals. From this pair of equations it is clear that, in the case of binary PSK, there is only one basis function of unit energy, namely, Φ1(t) = √2 / Tb cos(2πfct) 0 ≤ t < Tb (3) Then we may express the transmitted signals S1(t) and S2(t) in terms of Φ1(t) as follows: S1(t) = √ Eb Φ1(t) 0 ≤ t < Tb (4) S2(t) = -√ Eb Φ1(t) 0 ≤ t < Tb (5) 102/129 Matlab Source code for Binary Phase shift Keying clear ;close all;pack; t=[0:1/1000:1]; no=10000; numbits=input('Enter the number of data samples:'); %numbits=5; fc=1;tb=1; snr=[];ber=[]; carrier=cos(2*pi*fc*t);basis=sqrt(2/tb)*carrier; for eb=1:numbits mod=[];rxcode=[];demod=[]; data=randint(1,numbits);% Generating the bits of input sequence data(find(data==0))=-1; % converting polar form for j=1:numbits mod=[mod sqrt(eb)*data(j)*basis]; end Noise= sqrt(no)*randn(1,length(mod));% creating Noise recdcode=mod+Noise;% adding Noise for i=1:numbits % Product modultion with basis function demod=[demod sqrt(eb)*recdcode(:,(i-1)*length(t)+1:i*length(t)).*basis]; % ingration with time tb i.e sumation rxcode=[rxcode sum(demod(:,(i-1)*length(t)+1:i*length(t)))]; end % Decision Device rxcode(find(rxcode>=0))=1;rxcode(find(rxcode<0))=-1; %finding out the error error(eb,:)=data-rxcode;numerror=nnz(error(eb,:)); %finding out the bit error ber=[ber numerror/numbits]; %finding out the SNR sigpower=cov(mod);snr=[snr 10*log10(sigpower/sqrt(no))]; end dB=[]; for k=1:numbits Pb=[]; dB=[dB snr(k)]; SNR = 10.^ (dB ./ 10); Pb = [Pb 0.5*erfc(sqrt(SNR))]; % problity of error end t1=[0:1/1000:numbits]; figure(1); subplot(3,1,1);stem(data(1:numbits)); title('Wave pattern in BPSK');ylabel('data'); carr=cos(2*pi*fc*t1); subplot(3,1,2);plot(t1,carr);axis([0 numbits-1 -1 1]);ylabel('carrier'); subplot(3,1,3);plot(t1(1:numbits*1000),mod(1:numbits*1000)); axis([0 numbits-1 -5 5]);ylabel('transmitted code'); figure(2); subplot(2,1,1);stem(data(1:numbits));ylabel('transmitted code'); subplot(2,1,2);stem(rxcode(1:numbits));ylabel('received code'); figure(3);semilogy (dB(1:numbits), Pb(1:numbits)); grid on;xlabel('SNR (dB)');ylabel('Bit error rate'); title('Performance of antipodal signal over AWGN channel'); 103/129 FIGURE 1 Signal-space diagram for coherent binary PSK system. The waveforms depicting the transmitted signals S1(t) and S2(t) , displayed in the inserts, assume nc = 2. A coherent binary PSK system is therefore characterized by having a signal space that is one-dimensional (i.e., N = 1), with a signal constellation consisting of two message points (i.e., M = 2). The coordinates of the message points are S11 = ∫ S1(t) Φ1(t) dt (6) = + √Eb And S21 = ∫ S1(t) Φ1(t) dt (7) = - √Eb The message point corresponding to S1(t) is located at S11=+√Eb, and the message point corresponding to S2(t) is located at S21 =-√Eb Figure 1 displays the signal-space diagram for binary PSK. This figure also includes two inserts, showing example waveforms of antipodal signals representing S1(t) and S2(t). Note that the constellation of Figure 1 has minimum average energy. Generation and Detection of Coherent Binary PSK Signals To generate a binary PSK signal, we see from Equations (1)-(3) that represent the input binary sequence in polar form with symbols 1 and 0 represented by constant amplitude levels of + √Eb and - √Eb, respectively. This signal transmission encoding is performed by a polar nonreturn- to-zero (NRZ) level encoder. The resulting binary wave and a sinusoidal carrier Φ1(t) whose 104/129 Plot for polar input sequence, carrier and BPSK wave pattern 1 2 3 4 5 6 7 8 9 10 -1 -0.5 0 0.5 1 Wave pattern in BPSK data 0 1 2 3 4 5 6 7 8 9 -1 -0.5 0 0.5 1 carrier 0 1 2 3 4 5 6 7 8 9 -5 0 5 transmitted code Plot for Transmitted code and received code 1 2 3 4 5 6 7 8 9 10 -1 -0.5 0 0.5 1 transmitted code 1 2 3 4 5 6 7 8 9 10 -1 -0.5 0 0.5 1 received code 105/129 frequency fc = (nc / Tb) for some fixed integer nc, are applied to a product modulator, as in Figure 2a. The carrier and the timing pulses used to generate the binary wave are usually extracted from a common master clock. The desired PSK wave is obtained at the modulator output. To detect the original binary sequence of 1s and 0s, we apply the noisy PSK signal x(t) (at the channel output) to a correlator, which is also supplied with a locally generated coherent reference signal Φ1(t) ,as in Figure 2b. The correlator output, x1, is compared with a threshold of zero volts. If x1 > 0, the receiver decides in favor of symbol 1. On the other hand, if x1 < 0, it decides in favor of symbol 0. If x1 is exactly zero, the receiver makes a random guess in favor of 0 or 1 Algorithm Transmitter or Modulation � Generate the no of Random sequence bits as input given � Generate the carrier and basis function � Convert the binary sequence into polar non-return-to-zero (NRZ) level. i.e., for Symbol 1 for 1 & symbol zero -1 � To generate the modulated output multiply basis function with input bit sequence. � Generate the random noise signal � Add the noise with modulated signal � Plot the polar input data , carrier and modulated carrier Receiver or Demodulation � The Received signal is multiply with the same basis function. � sum the multiplied signal at the Tb interval (i.e. integration ) � Compare the summation data for > 0 or < 0. If >0 make Symbol 1 and < 0 Make Symbol 0. � Check the Error by transmitted bit – received bit. � Calculate the SNR � Calculate the bit Error Rate ( error/ no of bit) � Plot the SNR verses BER ( Bit Error Rate) � Plot the transmitted and received bit 106/129 Plot for BER verses SNR -20 -18 -16 -14 -12 -10 -8 10 -0.48 10 -0.46 10 -0.44 10 -0.42 10 -0.4 10 -0.38 10 -0.36 SNR (dB) Bit error rate Performance of antipodal signal over AWGN channel 107/129 Inference • The simple BPSK transmitter has only polar converter and product modulator. • The simple receiver has correlator (product modulator with integrator) and Decision device. • The BER and SNR depend on the Channel noise. Result The Binary phase shift keying Transmitter and Receiver understood and analyzed and plot the BER, SNR performance. 108/129 Pure Aloha Network Configuration Screenshot: 109/129 Analysis of MAC Protocol - ALOHA Aim: To study and analysis of network MAC protocol - Aloha Software Required: Netsim 2.0 version Software Theory: Introduction Protocols are strict languages used for communicating reliably between two compatible nodes. Each node must know how to interpret and send signals in accordance with the rules of a particular protocol. One of the most basic protocol is the Aloha Protocol. It is used to solve the channel allocation problem. It uses ground based radio broad casting, the basic idea is applicable to any system in which un-coordinated users are competing for use of a single shared channel. Pure Aloha With Pure Aloha station are allowed to access the channel whenever they have data to transmit and then waits for an acknowledgement is not received with in a short amount of time. The station assume that another station has also transmitted simultaneously causing a collision in which the combined transmission were distorted so that the receiving station did not hear them and did not return an acknowledgement and detecting a collision both transmission station would choose a random back of time and then retransmit their packet with a good probability of success however as traffic increased on the aloha channel, the collision rate would rapidly increases as well. 110/129 Simulated Output Screenshots: Frames / Frame time: 0.25 Frames / Frame time: 0.5 Frames / Frame time: 0.75 Frames / Frame time: 1.0 111/129 Network Performance Parameter Payload Delivered (Bytes) Total number of data bytes in successful frames transmitted in network. Overhead (Bytes) The total number of overhead transmitted with successful data bytes in the network during simulation Queuing time (Micro second) This is calculated for successful frames. Medium Access Time (Micro second) This is calculated for successful frames. Transmission Time (Micro second) This is calculated for successful frames. Dropped Frame Number of frames dropped at the source without transmission due to retransmission due to retransmission limit Error Frames The Number of frames discarded due to error. Total Attempts Sum of all the attempts made by nodes in the network to transmit data successfully. Successful Attempt Sum of all the attempts made by nodes in the network to transmit data frames successfully. Collision Counts Total number of collision in the network. Defer Counts Total number of times the node have postponed transmission in the network. 112/129 Frames / Frame Time Vs Throughput Frames / Frame Time Vs Mean Delay Serial no. Transmitting nodes Frames / frame time (second) Throughput(%) Mean Delay (micro second) 1 3 0.25 15.00118 7564438.521 2 4 0.5 13.47939 31265274.18 3 5 0.75 13.11574 51190881.89 4 6 1 12.53094 112968871 11 11.5 12 12.5 13 13.5 14 14.5 15 15.5 0.25 0.5 0.75 1 Frames/frame time(sec) Throughput(%) 0 20000000 40000000 60000000 80000000 100000000 120000000 0.25 0.5 0.75 1 Frames/frame time (sec) Mean Delay (us) 113/129 Simulation Time (Micro second) Time for transmitting all the data frames generated also means the time for draining the queue of all nodes. Data Frames Generated Total number of data frames generated by all nodes put together. Normalized Throughput Also called as good put. Fraction of link’s capacity devoted to carrying non retransmitted frames excluding bytes due to protocol overhead, collision and retransmission. Normalized Throughput (%) = (payload delivered x byte time x 100) /simulation end time Byte Time (Micro second) Time taken for a byte to reach the destination. This value depends on the data rate capacity of physical medium Throughput (%) Fraction of links capacity devoted to carrying frames. Throughput (%) = ((payload delivered + overheads) x Byte time)/simulation end time x 100 Mean Delay Mean time a frame waits at a station before being successfully transmitted (queuing time and medium access time and transmission time per frame). Formula Mean Delay = Queuing time + Medium access time + Transmission time in μsec/Frame (pay load delivered / maximum data size performance) Response Time (μ sec) Sum of medium access time and transmission time per frame. Formula Response Time = medium Access time + Transmission time (Payload delivered / maximum data size performance) 114/129 115/129 Probability of success Ratio of successful data transmission to overall transmission Formula Probability of success = successful attempt / total attempt. Average attempt The average attempt made by a node to successfully transmit a frame in the network. Formula Average attempt = (total attempt x slot time) / simulation end time Procedure 1. Run NetsimVer2.0 Software. Select Simulation – New – LAN – ALOHA option in the menu. 2. Select the nodes from the left side menu and drag it & place it in the workspace provided in the software. 3. Place minimum of 2nodes and maximum of 10nodes in the workspace. 4. Select a node and right click button in the mouse and select properties option. 5. Set the Configuration of the node as the Frames / frame time to 0.25 Sec. 6. Similarly set the Frames /frame time to 0.25 Sec to all nodes. 7. Run the simulation, save the simulation output to a file. 8. Repeat the same above procedure for different frames/ frame times 0.5, 0.75 & 1. 9. Save all the simulation output to files for all option. 10. Using the data stored in the simulation result file, Plot a graph by taking the Frame / frame time in X-axis & Throughput (%) in Y-axis. 11. Similarly Plot a graph by taking Frame / frame in X-axis & Mean delay in Y- axis. 116/129 117/129 Result Successfully studied and simulated the network MAC protocol - Pure Aloha. Inference The Following points are observed from the graph i. When then Frame / frame time (data rate) increases, the throughput decreases drastically. ii. The mean delay is also increased lot with the increase in Frame / frame time. Thus by considering the above points we can conclude pure aloha has low efficiency event though the protocol is simple. Hence the protocol is not suitable for high speed networking and it is suitable for low speed network with less number of nodes. 118/129 CSMA/CD Network Configuration Screenshot: 119/129 Analysis of MAC Protocol – CSMA/CD Aim: To study the theory of MAC Protocol CSMA/CD and analyze network performance of various simulations. Software Required: Netsim 2.0 version Software Theory: Carrier Sense Multiple Access (CSMA) Ethernet uses a refinement of ALOHA, known as Carrier Sense Multiple Access (CSMA), which improves performance when there is a higher medium utilisation. When a NIC has data to transmit, the NIC first listens to the cable (using a transceiver) to see if a carrier (signal) is being transmitted by another node. The individual bits are sent by encoding them with a 10 (or 100 MHz for Fast Ethernet) clock using Manchester encoding. Data is only sent when no carrier is observed (i.e. no current present) and the physical medium is therefore idle. Any NIC which does not need to transmit, listens to see if other NICs have started to transmit information to it. 120/129 Simulated Output Screenshots: Frames / Frametime: 0.25 Frames / Frametime: 0.5 Frames / Frametime: 0.75 Frames / Frametime: 1.0 121/129 However, this alone is unable to prevent two NICs transmitting at the same time. If two NICs simultaneously try transmit, then both could see an idle physical medium (i.e. neither will see the other's carrier signal), and both will conclude that no other NIC is currently using the medium. In this case, both will then decide to transmit and a collision will occur. The collision will result in the corruption of the frame being sent, which will subsequently be discarded by the receiver since a corrupted Ethernet frame will (with a very high probability) not have a valid 32-bit MAC CRC at the end. Collision Detection (CD) A second element to the Ethernet access protocol is used to detect when a collision occurs. When there is data waiting to be sent, each transmitting NIC also monitors its own transmission. If it observes a collision , it stops transmission immediately and instead transmits a 32-bit jam sequence. The purpose of this sequence is to ensure that any other node which may currently be receiving this frame will receive the jam signal in place of the correct 32-bit MAC CRC, this causes the other receivers to discard the frame due to a CRC error. 122/129 Frames/Frame Vs Mean Delay Frames/Frame time 0.25 0.5 0.75 1 Mean Delay (sec) 0.00029 0.8388300 0.84819 0.147324 Frames/Frametime Vs Mean Delay (sec) 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 0.25 0.5 0.75 1 Frames/Frametime Mean Delay (sec) Frames/Frame Vs Throughput Frames/Frametime Vs Throughput (%) 0 20 40 60 80 100 120 0.25 0.5 0.75 1 Frames/Frametime Throughput (%) Frames/Frame time 0.25 0.5 0.75 1 Throughput (%) 50.74352 75.03895 87.74781 99.20868 123/129 Procedure: 1. Using Netsim software, configure the network with number of nodes and switches as shown in the below screen shot. 2. For each simulation set different value of frames/frame time by changing the properties of nodes and note down all the readings. 3. Now network should be configured by clicking configure button in simulator. 4. Then simulate network by using simulate option in the simulator and take screen shots for output. 5. Graphs should be drawn between i. Frames / Frametime Vs Mean Delay (sec) ii. Frames / Frametime Vs Throughput (%) Result: Theory of MAC protocol CSMA/CD was studied and the network performance is also analyzed. Inference: i. The variation in the mean delay for the number of frames generated is approximately linear for both protocols. ii. The throughput (%) of CSMA/CD protocol is remarkably higher than CSMA protocol. iii. When two or more NICs attempt to transmit at the same time, the performance of Ethernet is less predictable. The fall in utilisation and throughput occurs because some bandwidth is wasted by collisions and back-off delays. 124/129 Token Ring Network Configuration Screenshot: 125/129 ANALYSIS OF MAC PROTOCOL – TOKEN RING Aim: To study the network performance of MAC Protocol Token ring using NETSIM simulation Software Required: Netsim 2.0 version Software Theory: The token ring technique is based on the use of a small frame, called a token that circulates when all stations are idle. A station wishing to transmit must wait until it detects a token passing by. It then seizes the token by changing one bit in the token, which transforms it from a token into a start-of-frame sequence for a data frame. The station then appends and transmits the remainder of the fields needed to construct a data frame. When a station seizes a token and begins to transmit a data frame, there is no token on the ring, so other stations wishing to transmit must wait. The frame on the ring will make a round trip and be absorbed by the transmitting station. The transmitting station will insert a new token on the ring when both of the following conditions have been met: • The station has completed transmission of its frame. • The leading edge of the transmitted frame has returned (after a complete circulation of the ring) to the station. If the bit length of the ring is less than the frame length, the first condition implies the second if not, a station could release a free token after it has finished transmitting but before it begins to receive its own transmission. The second condition is not strictly necessary, and is relaxed under certain circumstances. The advantage of imposing the second condition is that it ensures that only one data frame at a time may be on the ring and that only one station at a time may be transmitting, thereby simplifying error-recovery procedures. 126/129 Simulated Output Screenshots: Frames / Frame time: 0.25 Frames / Frame time: 0.5 Frames / Frame time: 0.75 Frames / Frame time: 1.0 127/129 Once the new token has been inserted on the ring, the next station downstream with data to send will be able to seize the token and transmit. Figure illustrates the technique. In the example, A sends a packet to C, which receives it and then sends its own packets to A and D. Note that under lightly loaded conditions, there is some inefficiency with token ring because a station must wait for the token to come around before transmitting. However, under heavy loads, which is when it matters, the ring functions in a round-robin fashion, which is both efficient and fair. To see this, consider the configuration in Figure. After station A transmits, it releases a token. The first station with an opportunity to transmit is D. If D transmits, it then releases a token and C has the next opportunity, and so on. The principal advantage of token ring is the flexible control over access that it provides. In the simple scheme just described, the access if fair. As we shall see, schemes can be used to regulate access to provide for priority and for guaranteed bandwidth services. The principal disadvantage of token ring is the requirement for token maintenance. Loss of the token prevents further utilization of the ring. Duplication of the token can also disrupt ring operation. One station must be selected as a monitor to ensure that exactly one token is on the ring and to ensure that a free token is reinserted, if necessary. Procedure: 1. Using Netsim Software, Token Ring protocol is configured 2. The Network environment is opened 3. Nodes are selected, dragged, placed and interconnected in the Network environment 4. The properties of Individual node Vs Frame size (200, 400, 400 & 500 kbs) are configured by Right clicking the nodes 5. The Network environment is simulated. 6. The network performance of environment is obtained. 7. The effect of Frame size on Mean delay and Throughput in % are plotted 128/129 Frames/Frame Time Vs Throughput Frames/Frame Time Vs Mean Delay Frames Throughput Mean Delay 0.25 96.839839 104182.266 0.5 99.97791 4968629.547 0.75 99.977887 9975155.298 1 99.835105 99.977931 99.974 99.9745 99.975 99.9755 99.976 99.9765 99.977 99.9775 99.978 99.9785 99.979 99.9795 0.25 0.5 0.75 1 Frames/frame time(sec) Throughput(%) 0 20000000 40000000 60000000 80000000 100000000 120000000 0.25 0.5 0.75 1 Frames/Frame time(sec) Mean Delay(us) 129/129 Result: The characteristics of Network environment with different component properties i.e. the effect of Frame size on Mean delay and Throughput in % of MAC Protocol Token Ring is analyzed using NETSIM Inference: When we increase the frame/frames time, throughout is decreased. Frame/frames time is indirectly proportional to throughput. When we increase the frame/frames time mean delay is increased. Frame/frames time is directly proportional to mean delay.