TELEMETRY ASSIGNMENT

SUBMITTED BY: SINGH KUSHAGRA HARILAL EI- H 0909732109

Q.1 WHAT IS MODEM? EXPLAIN.

A modem (modulator-demodulator) is a device that modulates an analog carrier signal to encode digital information, and also demodulates such a carrier signal to decode the transmitted information. The goal is to produce a signal that can be transmitted easily and decoded to reproduce the original digital data. Modems can be used over any means of transmitting analog signals, from light emitting diodes to radio. The most familiar example is a voice modem that turns the digital data of a personal computer into modulated electrical signals in the voice frequency range of a telephone channel. These signals can be transmitted over telephone lines and demodulated by another modem at the receiver side to recover the digital data. Modems are generally classified by the amount of data they can send in a given unit of time, usually expressed in bits per second (bit/s, or bps), or bytes (B/s). Modems can alternatively be classified by their symbol rate, measured in baud. The baud unit denotes symbols per second, or the number of times per second the modem sends a new signal. For example, the ITU V.21 standard used audio frequency shift keying, that is to say, tones of different frequencies, with two possible frequencies corresponding to two distinct symbols

(or one bit per symbol), to carry 300 bits per second using 300 baud. By contrast, the original ITU V.22 standard, which was able to transmit and receive four distinct symbols (two bits per symbol), handled 1,200 bit/s by sending 600 symbols per second (600 baud) using phase shift keying.

Q.2 EXPLAIN PRINCIPLES OF QAM.

Quadrature amplitude modulation (QAM) is both an analog and a digital modulation scheme. It conveys two analog message signals, or two digital bit streams, by changing (modulating) the amplitudes of two carrier waves, using the amplitude-shift keying (ASK) digital modulation scheme or amplitude modulation (AM) analog modulation scheme. The two carrier waves, usually sinusoids, are out of phase with each other by 90 and are thus called quadrature carriers or quadrature components hence the name of the scheme. The modulated waves are summed, and the resulting waveform is a combination of both phase-shift keying (PSK) and amplitude-shift keying (ASK), or (in the analog case) of phase modulation (PM) and amplitude modulation. In the digital QAM case, a finite number of at least two phases and at least two amplitudes are used. PSK modulators are often designed using the QAM principle, but are not considered as QAM since the amplitude of the modulated carrier signal

is constant. QAM is used extensively as a modulation scheme for digital telecommunication systems. Arbitrarily high spectral efficiencies can be achieved with QAM by setting a suitable constellation size, limited only by the noise level and linearity of the communications channel.[1] QAM modulation is being used in optical fiber systems as bit rates increase; QAM16 and QAM64 can be optically emulated with a 3-pathinterferometer.

Q.3 EXPLAIN DIFFERENT MODEM PROTOCOLS.

Interface Speed and Connection Speed When you make a dialup connection, there are (at least) three separate components to the connection. Assuming you are dialing up from a PC, there is the connection between your PC and your (originating) modem, the connection between the two modems, and the connection between the "other" (answering) modem and whatever device it is attached to. Each of part of the connection can be running at a different speed: 1.LOCAL INTERFACE SPEED: The speed used on the connection between your PC (or terminal or workstation) and your modem.

2.CONNECTION SPEED: The speed of the connection between the two modems, based on the modulation technique that they negotiate with each other. 3.REMOTE INTERFACE SPEED: The speed of the connection between the remote (answering) modem and the terminal server. Speed Buffering When originating a call, some modems change their interface speed to match the negotiated connection speed automatically. When that happens, your communications software must also change its speed at the same time. For example, if you dial at 9600 bps, but the remote modem answers at 1200 bps, your modem will print a message like: CONNECT 1200 which your communication software takes as a signal to change its interface speed to 1200 bps before attempting to go "online" with the remote computer or service. Most modern modems can be configured to fix their interface speed at a given value, rather than change it according to the connection speed. This is desirable when using data compression. In this case, the CONNECTION SPEED (or MODULATION SPEED) between the two modems is different from the INTERFACE SPEED between the modem and the computer. The modem performs the speed conversion between its telephone side and its data side, and your communications

software must be configured to IGNORE the speed given in the CONNECT message.

Error Correction After the modems have agreed on a modulation technique, they might also try to negotiate an error-detection and -correction method: MNP Level 1, 2, 3, or 4 ITU-T V.42 = LAPM Telebit PEP (proprietary) US Robotics HST (proprietary) When modems' initial error-control methods do not agree, automatic fallback is usually as follows: V.42 ->. MNP 4 ->. MNP 3 ->. MNP 2 ->. MNP 1 ->. none When PEP, HST, or other proprietary methods are involved, special configuration settings are needed on the modems to specify the fallback sequence. Please note that no connection can ever be free of errors. The error correction technique used between the modems might be extremely effective, but it is not foolproof. More to the point, the connections between the modems and the computers are not errorcorrected, nor or the data paths within the computers. Thus it is still quite common to experience data loss or corruption, even on an

error-corrected modem connection. Common causes include: buffer overflows (often due to a lack of adequate flow control between the modem and the computer -- see below), interrupt conflicts, loose connectors, and malfunctioning devices. Beware of RPI modems. They do NOT perform error correction themselves, but rely on external software to do it. Most software does not. These modems are NOT SUPPORTED at Columbia University. Data Compression Modems may incorporate data compression methods to increase the effective throughput of your data beyond the actual connection speed. Compression is possible only if (a) error correction is also being done, and (b) the interface speed between the computer and the modem is higher than the connection speed between the two modems. MNP Level 3 108% efficiency by removing start & stop bits (synchronous) MNP Level 4 120% efficiency by optimizing modem-to-modem protocol MNP Level 5 True compression on top of Level 4, efficiency depends on data V.42bis True compression, efficiency depends on data

Telebit PEP Proprietary, characteristics unknown US Robotics HST Proprietary, characteristics unknown Effectiveness of MNP 5 and V.42bis compression vary between 0% and 400% or higher, depending on the nature of the data. Compression fallback: V.42bis ->. MNP 5 ->. None When PEP or HST is involved, special configuration settings are needed on the modem to specify how these fit into the fallback sequence. Again, beware of RPI modems. They do not do compression themselves, but rely on external software to do it. Flow Control "Flow control" is the method by which one device can control the rate at which another device sends data to it. There are various methods of flow control. The two most commonly used in dialup data communication are:

RTS/CTS (Request To Send / Clear To Send) "hardware" flow control is the most effective method. It uses special wires in the cable (or, in the case of an internal modem, special signals on the edge-connector), separate from the data wires, to control the flow of data. It

is used between two devices that are immediately connected, such as a computer and a modem. XON/XOFF "software" flow control is less effective and more risky because it mixes flow control characters (Control-S and Control-Q) with the data. These characters are subject to delay, loss, and corruption, and also cause transparency problems with host applications like EMACS. Software flow control should be used only if hardware flow control is unavailable.

When using error correction or compression, or modems that are capable of retraining, it is essential to enable an effective form of flow control between each modem and the computer (or terminal, or other device) it is immediately connected to. Without effective flow control, data will be lost when one device sends data faster than the other one can receive it. Flow control between the two modems is handled by the underlying error modem-to-modem correction protocol: MNP or V.42. If there is no underlying error-correction protocol, then there can be no flow control between the modems and therefore no protection against data loss EVEN IF there is flow control between the modem and the computer. This applies, in particular, to RPI modems.

Q.4 What are the different multiplexing techniques used for communication technology. Explain the working of FDM in details with suitable diagrams & how it differs from TDM.
Space-division multiplexing

Main article: space-division multiplexing

In wired communication, space-division multiplexing simply implies different point-to-point wires for different channels. Examples include an analogue stereo audio cable, with one pair of wires for the left channel and another for the right channel, and a multipart telephone cable. Another example is a switched star network such as the analog telephone access network (although inside the telephone exchange or between the exchanges, other multiplexing techniques are typically employed) or a switched Ethernet network. A third example is a mesh network. Wired space-division multiplexing is typically not considered as multiplexing. In wireless communication, space-division multiplexing is achieved by multiple antenna elements forming a phased array antenna. Examples are multiple-input and multiple-output (MIMO),

single-input and multiple-output (SIMO) and multiple-input and single-output (MISO) multiplexing. For example, a IEEE 802.11n wireless router with N antennas makes it possible to communicate with Multiplexed channels, each with a peak bit rate of 54 Mbit/s, thus increasing the total peak bit rate with a factor N. Different antennas would give different multi-path propagation (echo) signatures, making it possible for digital signal processing techniques to separate different signals from each other. These techniques may also be utilized for space diversity (improved robustness to fading) or beam forming (improved selectivity) rather than multiplexing. [edit]Frequency-division multiplexing

Main article: Frequency-division multiplexing

Frequency-division multiplexing (FDM): The spectrum of each input signal is shifted to a distinct frequency range. Frequency-division multiplexing (FDM) is inherently an analog technology. FDM achieves the combining of several digital signals into one medium by sending signals in several distinct frequency ranges over that medium. One of FDM's most common applications is cable television. Only one cable reaches a customer's home but the service provider can send multiple television channels or signals simultaneously over that cable to all subscribers. Receivers must tune

to the appropriate frequency (channel) to access the desired signal.[1] A variant technology, called wavelength-division multiplexing (WDM) is used in optical communications. Time-division multiplexing

Main article: Time-division multiplexing

Time-division multiplexing (TDM). Time-division multiplexing (TDM) is a digital (or in rare cases, analog) technology. TDM involves sequencing groups of a few bits or bytes from each individual input stream, one after the other, and in such a way that they can be associated with the appropriate receiver. If done sufficiently quickly, the receiving devices will not detect that some of the circuit time was used to serve another logical communication path. Consider an application requiring four terminals at an airport to reach a central computer. Each terminal communicated at 2400 bit/s, so rather than acquire four individual circuits to carry such a low-speed transmission, the airline has installed a pair of multiplexers. A pair of 9600 bit/s modems and one dedicated analog communications circuit from the airport ticket desk back to the airline data center are also installed.[1] Polarization-division multiplexing

Main article: Polarization-division multiplexing

Polarization-division multiplexing uses the polarization of electromagnetic radiation to separate orthogonal channels. It is in practical use in both radio and optical communications, particularly in 100 Gbit/s per channel fiber optic transmission systems. Orbital angular momentum multiplexing

Main article: Orbital angular momentum multiplexing

Orbital angular momentum multiplexing is a relatively new and experimental technique for multiplexing multiple channels of signals carried using electromagnetic radiation over a single path.[2] It can potentially be used in addition to other physical multiplexing methods to greatly expand the transmission capacity of such systems. As of 2012 it is still in its early research phase, with small-scale laboratory demonstrations of bandwidths of up to 2.5 Tbit/s over a single light path.[3] Code-division multiplexing

Main articles: Spread spectrum and Code division multiplexing

Code division multiplexing (CDM) or spread spectrum is a class of techniques where several channels simultaneously share the same frequency spectrum, and this spectral bandwidth is much higher than the bit rate or symbol rate. One form is frequency hopping, another is direct sequence spread

spectrum. In the latter case, each channel transmits its bits as a coded channel-specific sequence of pulses called chips. Number of chips per bit, or chips per symbol, is the spreading factor. This coded transmission typically is accomplished by transmitting a unique time-dependent series of short pulses, which are placed within chip times within the larger bit time. All channels, each with a different code, can be transmitted on the same fiber or radio channel or other medium, and asynchronously demultiplexed. Advantages over conventional techniques are that variable bandwidth is possible (just as in statistical multiplexing), that the wide bandwidth allows poor signal-to-noise ratio according to Shannon-Hartley theorem, and that multi-path propagation in wireless communication can be combated by rake receivers. Code Division Multiplex techniques are used as an channel access scheme, namely Code Division Multiple Access (CDMA), e.g. for mobile phone service and in wireless networks, with the advantage of spreading intercell interference among many users. Confusingly, the generic term Code Division Multiple access sometimes refers to a specific CDMA based cellular system defined by Qualcomm. Another important application of CDMA is the Global Positioning System (GPS). FREQUENCY DIVISION MULTIPLEXING

In telecommunications, frequency division multiplexing (FDM) is a technique by which the total bandwidth available in a communication medium is divided into a series of nonoverlapping frequency sub-bands, each of which is used to carry a separate signal. This allows a single transmission medium such as a cable or optical to be shared by many signals. An example of a system using FDM is cable television, in which many television channels are carried simultaneously on a single cable. FDM is also used by telephone systems to transmit multiple telephone calls through high capacity trunk lines, communications satellites to transmit multiple channels of data on uplink and downlink radio beams, and broadband DSL modems to transmit large amounts of computer data through twisted pair telephone lines, among many other uses.

Q.5 5 Explain the working of a TDM-PCM system with a neat block diagram, also illustrate all the components of the block. Explain how the TDM-PCM system is used for A.M and F.M?

In conventional PCM, the analog signal may be processed (e.g., by amplitude compression) before being digitized. Once the signal is digitized, the PCM signal is usually subjected to further processing (e.g., digital data compression). PCM with linear quantization is known as Linear PCM (LPCM).[9] Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog domain as part of the analog-to-digital process; newer implementations do so in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques.

DPCM encodes the PCM values as differences between the current and the predicted value. An algorithm predicts the next sample based on the previous samples, and the encoder stores only the difference between this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same information. For audio, this type of encoding reduces the number of bits required per sample by about 25% compared to PCM. Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio.

Delta modulation is a form of DPCM which uses one bit per sample.

In telephony, a standard audio signal for a single phone call is encoded as 8,000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either -law (mulaw) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 12 or 13-bit linear PCM sample number is mapped into an 8bit value. This system is described by international standard G.711. An alternative proposal for a floating point representation, with 5-bit mantissa and 3-bit radix, was abandoned. Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm is used to map a series of 8-bit -law or A-law PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard. Later it was found that even further compression was possible and additional standards were published. Some of these international standards describe systems and ideas which are covered by privately owned patents and thus use of these standards requires payments to the patent holders.

Some ADPCM techniques are used in Voice over IP communications. PCM can be either return-to-zero (RZ) or non-returnto-zero (NRZ). For a NRZ system to be synchronized using in-band information, there must not be long sequences of identical symbols, such as ones or zeroes. For binary PCM systems, the density of 1symbols is called ones-density.[10] Ones-density is often controlled using precoding techniques such as Run Length Limited encoding, where the PCM code is expanded into a slightly longer code with a guaranteed bound on ones-density before modulation into the channel. In other cases, extra framing bits are added into the stream which guarantee at least occasional symbol transitions. Another technique used to control ones-density is the use of a scrambler polynomial on the raw data which will tend to turn the raw data stream into a stream that looks pseudo-random, but where the raw stream can be recovered exactly by reversing the effect of the polynomial. In this case, long runs of zeroes or ones are still possible on the output, but are considered unlikely enough to be within normal engineering tolerance. In other cases, the long term DC value of the modulated signal is important, as building up a DC offset will tend to bias detector circuits out of their operating range. In this case special measures are taken to keep a count of the cumulative DC

offset, and to modify the codes if necessary to make the DC offset always tend back to zero. Many of these codes are bipolar codes, where the pulses can be positive, negative or absent. In the typical alternate mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to generate special symbols used for framing or other special purposes.

Q.6 Explain generation and propagation of A.M with suitable ckt diagram for each?
Amplitude modulation (AM) is a technique used in electronic communication, most commonly for transmitting information via a radio carrier wave. AM works by varying the strength of the transmitted signal in relation to the information being sent. For example, changes in signal strength may be used to specify the sounds to be reproduced by a loudspeaker, or the light intensity of television pixels. Contrast this with frequency modulation, in which thefrequency is varied, and phase modulation, in which the phase is varied. In the mid-1870s, a form of amplitude modulation initially called "undulatory currents"was the first method to successfully produce quality audio

over telephone lines. Beginning with Reginald Fessenden's audio demonstrations in 1906, it was also the original method used for audio radio transmissions, and remains in use today by many forms of communication"AM" is often used to refer to the mediumwave broadcast band (see AM radio).

Modulation methods

Anode (plate) modulation. A tetrode's plate and screen grid voltage is modulated via an audio transformer. The resistor R1 sets the grid bias; both the input and output are tuned circuits with inductive coupling. Modulation circuit designs may be classified as lowor high-level (depending on whether they modulate in a low-power domainfollowed by amplification for transmissionor in the high-power domain of the transmitted signal).[1] [edit]Low-level generation In modern radio systems, modulated signals are generated via digital signal processing (DSP). With DSP many types of AM modulation are possible with software control (including DSB with carrier, SSB suppressed-carrier and independent sideband, or ISB). Calculated digital samples are converted to voltages with a digital to analog converter, typically at a frequency less than the desired RFoutput frequency. The analog signal must then be shifted in frequency, and linearly amplified to the desired frequency and power level (linear amplification must be used to prevent modulation distortion).[2] This low-level method for AM is used in many Amateur Radio transceivers.[3] AM may also be generated at a low level, using analog methods described in the next section. [edit]High-level generation

High-power AM transmitters (such as those used for AM broadcasting) are based on high-efficiency class-D and class-E power amplifierstages, modulated by varying the supply voltage.[4] Older designs (for broadcast and amateur radio) also generate AM by controlling the transmitters final amplifier (generally a class-C, for efficiency) gain. The following types are for vacuum tube transmitters (but similar options are available with transistors):[5]

Plate modulation: In plate modulation, the plate voltage of the RF amplifier is modulated with the audio signal. The audio power requirement is 50 percent of the RF-carrier power. Heising (constant-current) modulation: RF amplifier plate voltage is fed through a choke (high-value inductor). The AM modulation tube plate is fed through the same inductor, so the modulator tube diverts current from the RF amplifier. The choke acts as a constant current source in the audio range. This system has a low power efficiency. Control grid modulation: The operating bias and gain of the final RF amplifier can be controlled by varying the voltage of the control grid. This method requires little audio power, but care must be taken to reduce distortion. Clamp tube (screen grid) modulation: The screengrid bias may be controlled through a clamp

tube, which reduces voltage according to the modulation signal. It is difficult to approach 100-percent modulation while maintaining low distortion with this system.

Q.7 Explain the role of sample and hold ckt in a PAM-PCM connection system in details?
In electronics, a sample and hold (S/H, also "follow-and-hold"[1]) circuit is an analog device that samples (captures, grabs) the voltage of a continuously varying analog signal and holds (locks, freezes) its value at a constant level for a specified minimal period of time. Sample and hold circuits and related peak detectors are the elementary analog memory devices. They are typically used in analog-to-digital converters to eliminate variations in input signal that can corrupt the conversion process.[2] A typical sample and hold circuit stores electric charge in a capacitor and contains at least one fast FET switch and at least one operational amplifier.[1] To sample the input signal the switch connects the capacitor to the output of a buffer amplifier. The buffer amplifier charges or discharges the capacitor so that the voltage across the capacitor is practically equal, or proportional to, input voltage. In hold mode the switch

disconnects the capacitor from the buffer. The capacitor is invariably discharged by its own leakage currents and useful load currents, which makes the circuit inherently volatile, but the loss of voltage (voltage drop) within a specified hold time remains within an acceptable error margin. In the context of LCD screens, it is used to describe when a screen samples the input signal, and the frame is held there without redrawing it. This does not allow the eye to refresh and leads to blurring during motion sequences, also the transition is visible between frames because the backlight is constantly illuminated, adding to display motion blur

Q.8 Draw and explain the parts PCM receiver system used for telemetry?
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form for digital audio in computers and various Blu-ray, DVD and Disc formats, as well as

other uses such as digital telephone systems. A PCM stream is a digital representation of an analog signal, in which the magnitude of the analog signal is sampled regularly at uniform intervals, with each sample being quantized to the nearest value within a range of digital steps. PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that each sample can take.

Q.9 Explain the working principle of differential PCM with proper block diagram ,explain how signal encoding and decoding are being done by a DPCM in details .
Differential pulse-code modulation (DPCM) is a signal encoder that uses the baseline of pulse-code modulation (PCM) but adds some functionalities based on the prediction of the samples of the signal. The input can be an analog signal or a digital signal. If the input is a continuous-time analog signal, it needs to be sampled first so that a discrete-time signal is the input to the DPCM encoder.

Option 1: take the values of two consecutive samples; if they are analog samples, quantize them; calculate the difference between the first one and the next; the output is

the difference, and it can be further entropy coded. Option 2: instead of taking a difference relative to a previous input sample, take the difference relative to the output of a local model of the decoder process; in this option, the difference can be quantized, which allows a good way to incorporate a controlled loss in the encoding.

Applying one of these two processes, short-term redundancy (positive correlation of nearby values) of the signal is eliminated; compression ratios on the order of 2 to 4 can be achieved if differences are subsequently entropy coded, because the entropy of the difference signal is much smaller than that of the original discrete signal treated as independent samples. DPCM was invented by C. Chapin Cutler at Bell Labs in 1950; his patent includes both methods.

Q.10 What is the role of a delta modulator and delta demodulator in DPCM?
Delta modulation (DM or -modulation)is an analogto-digital and digital-to-analog signal conversion technique used for transmission of voice information where quality is not of primary importance. DM is the simplest form of differential pulse-code modulation (DPCM) where the difference between successive samples are encoded into n-bit data streams. In delta modulation, the transmitted data is reduced to a 1-bit data stream. Its main features are:

the analog signal is approximated with a series of segments each segment of the approximated signal is compared to the original analog wave to determine the increase or decrease in relative amplitude the decision process for establishing the state of successive bits is determined by this comparison only the change of information is sent, that is, only an increase or decrease of the signal amplitude from the previous sample is sent whereas a no-change condition causes the

modulated signal to remain at the same 0 or 1 state of the previous sample. To achieve high signal-to-noise ratio, delta modulation must use oversampling techniques, that is, the analog signal is sampled at a rate several times higher than the Nyquist rate. Derived forms of delta modulation are continuously variable slope delta modulation, delta-sigma modulation, anddifferential modulation. Differential pulse-code modulation is the super set of DM.

Q.11 Why a band pass filter is used in a demodulator chanel on receiving side of FDM telemetry system? Draw the schematic

diagram of the receiver side of the system and explain its operation?
A band-pass filter is a device that passes frequencies within a certain range and rejects (attenuates) frequencies outside that range. Optical band-pass filters are of common usage. An example of an analogue electronic bandpass filter is an RLC circuit (a resistor inductorcapacitor circuit). These filters can also be created by combining a low-pass filter with a high-pass filter.[1]

Bandpass is an adjective that describes a type of

filter or filtering process; it is frequently confused with passband, which refers to the actual portion of affected spectrum. Hence, one might say "A dual bandpass filter has two passbands." A bandpass signal is a signal containing a band of frequencies away from zero frequency, such as a signal that comes out of a bandpass filter.[2] An ideal bandpass filter would have a completely flat passband (e.g. with no gain/attenuation throughout) and would completely attenuate all frequencies outside the passband. Additionally, the transition out of the passband would be instantaneous in frequency. In practice, no bandpass

filter is ideal. The filter does not attenuate all frequencies outside the desired frequency range completely; in particular, there is a region just outside the intended passband where frequencies are attenuated, but not rejected. This is known as the filter roll-off, and it is usually expressed in dB of attenuation per octave or decade of frequency. Generally, the design of a filter seeks to make the roll-off as narrow as possible, thus allowing the filter to perform as close as possible to its intended design. Often, this is achieved at the expense of pass-band or stop-band ripple. The bandwidth of the filter is simply the difference between the upper and lower cutoff frequencies. The shape factor is the ratio of bandwidths measured using two different attenuation values to determine the cutoff frequency, e.g., a shape factor of 2:1 at 30/3 dB means the bandwidth measured between frequencies at 30 dB attenuation is twice that measured between frequencies at 3 dB attenuation. Outside of electronics and signal processing, one example of the use of band-pass filters is in the atmospheric sciences. It is common to band-pass filter recent meteorological data with a period range of, for example, 3 to 10 days, so that only cyclones remain as fluctuations in the data fields. In neuroscience, visual cortical simple cells were first shown by David Hubel and Torsten Wiesel to

have response properties that resemble Gabor filters, which are band-pass.

Q.12 How is a class C amplifier used to obtain frequency

multiplication? draw the schematic of three stage class C amplifier .using input ,collector current and output waveform ,demonstrate the multiplication property.how are the stages matched with each other?
In electronics, a frequency multiplier is an electronic circuit that generates an output signal whose output frequency is a harmonic (multiple) of its input frequency. Frequency multipliers consist of a nonlinear circuit that distorts the input signal and consequently generates harmonics of the input signal. A subsequent bandpass filter selects the desired harmonic frequency and removes the unwanted fundamental and other harmonics from the output. Frequency multipliers are often used in frequency synthesizers and communications circuits. It can be more economical to develop a lower frequency signal with lower power and less expensive devices, and then use a frequency multiplier chain to generate an

output frequency in the microwave or millimeter wave range. Some modulation schemes, such as frequency modulation, survive the nonlinear distortion without ill effect (but schemes such as amplitude modulation do not). Frequency multiplication is also used in nonlinear optics. The nonlinear distortion in crystals can be used to generate harmonics of laser light. A pure sinewave at frequency f has no harmonics. If it goes through a linear amplifier, the result continues to be pure (but may acquire a phase shift). If the sinewave is run through a stateless nonlinear circuit (transcribing function), the resulting distortion creates harmonics. The distorted signal can be described by a Fourier series in f.

The nonzero ck represent the generated harmonics. The Fourier coefficients are given by integrating over the fundamental period T:

These harmonics can be selected by a bandpass filter. The power in the distorted signal is spread across all the resulting harmonics.[1] An ideal halfwave rectifier, for example, has all

nonzero coefficients. An approximate circuit could use a diode. From a conversion efficiency standpoint, the nonlinear circuit should maximize the coefficient for the desired harmonic and minimize the others. Consequently, the transcribing function is often specially chosen. Easy choices are to use an even function to generate even harmonics or an odd function to for odd harmonics. See Even and odd functions#Harmonics. A full wave rectifier, for example, is good for making a doubler. To produce a times-3 multiplier, the original signal may be input to an amplifier that is over driven to produce nearly a square wave. This signal is high in 3rd order harmonics and can be filtered to produce the desired x3 outcome. YIG multipliers often want to select an arbitrary harmonic, so they use a stateful distortion circuit that converts the input sine wave into an approximate impulse train. The ideal (but impractical) impulse train generates an infinite number of (weak) harmonics. In practice, an impulse train generated by a monostable circuit will have many usable harmonics. YIG multipliers using step recovery diodes may, for example, take an input frequency of 1 to 2 GHz and produce outputs up

to 18 GHz.[2] Sometimes the frequency multiplier circuit will adjust the width of the impulses to improve conversion efficiency for a specific harmonic.

Q.13 Why is varactor diode is so popular in FM circuits? How does its capacitance vary with the amount of reverse bias?

An electronic circuit or device producing frequency modulation. This device changes the frequency of an oscillator in accordance with the amplitude of a modulating signal. If the modulation is linear, the frequency change is proportional to the amplitude of the modulating voltage. High-frequency oscillators usually employ either LC (inductance-capacitance) tuned circuits or piezoelectric crystals to establish the frequency of oscillation. This frequency can be controlled by changing the effective capacitance or inductance of the tuned circuit in accordance with the modulating signal. Practical circuits usually employ a varactor diode to change the oscillator in accordance with a modulating voltage. The oscillators in high-frequency electronic systems, such as frequency-modulating (FM) transmitters, usually employ piezoelectric crystals for precise control of the carrier frequency. These crystals are equivalent to a series LC tuned circuit with an extremely high Q. The crystal holder has a small capacitance which is in parallel with the crystal and therefore causes parallel resonance at a slightly higher frequency than the series resonant frequency of the crystal. The actual oscillator frequency is between these two resonant frequencies and is controllable by the parallel capacitance.

The junction capacitance of a semiconductor diode varies with the diode voltage, and a reverse-biased diode may be used to control the oscillator frequency to produce frequency modulation. Low-loss diodes designed for this service are known as varactor diodes and have trade names such as Varicaps or Epicaps. A basic varactor modulating scheme is shown in the illustration. In this circuit, the transistor that drives the varactor modulator provides reverse bias as well as the modulating voltage vm. The radio-frequency (rf) choke provides very high impedance at the oscillator frequency to isolate the transistor amplifier output impedance from the oscillator circuit but to allow the modulating signal to pass through with negligible attenuation. Only the frequencydetermining part of the oscillator is shown. The symbols Cc, Lc, and R represent the electrical equivalents of the compliance, mass, and loss, respectively, of the crystal; Ch is the crystalholder capacitance and Cb is a dc blocking capacitor

The capacitance of D1, of course, is controlled by two factors: a fixed dc bias and the modulating signal. In Fig b, the bias on D1 is set by the voltage divider which is made up of R1 and R2. Usually either R1 or R2 is made variable so that the center carrier frequency can be adjusted over a

narrow range. The modulating signal is applied through C3 and the RFC. The C3 is a blocking capacitor that keeps the DC bias out of the modulating signal circuits. The RFC is a radio frequency choke whose reactance is high at the carrier frequency to prevent the carrier signal from getting into the modulating signal circuits. The modulating signal derived from the microphone is amplified and applied to the modulator. As the modulating signal varies, it adds to or subtracts from the fixed bias voltage. Thus the effective voltage applied to D1 causes its capacitance to vary. This, in turn, produces a deviation of the carrier frequency as desired. A positive-going signal at point A adds to the reverse bias, decreasing the capacitance and increasing the carrier frequency. A negative-going signal at A subtracts from the bias, increasing the capacitance and decreasing the carrier frequency.

Q.14 What are the two basic discriminator used in the FM demodulator system? How do they differ in fuction?
There are several ways of demodulation depending on how parameters of the base-band signal are transmitted in the carrier signal, such as amplitude, frequency or phase. For example, for a signal modulated with a linear modulation, like AM (Amplitude Modulation), we can use a synchronous detector. On the other hand, for a signal modulated with an angular modulation, we must use an FM (Frequency Modulation) demodulator or a PM (Phase Modulation) demodulator. Different kinds of circuits perform these functions. Many techniquessuch as carrier recovery, clock recovery, bit slip, frame synchronization, rake receiver, pulse compression, Received Signal Strength Indication, error detection and correction, etc. -- are only performed by demodulators, although any specific demodulator may perform only some or none of these techniques. There are several common types of FM demodulator:

The quadrature detector, which phase shifts the signal by 90 degrees and multiplies it with the

unshifted version. One of the terms that drops out from this operation is the original information signal, which is selected and amplified. The signal is fed into a PLL and the error signal is used as the demodulated signal. The most common is a Foster-Seeley discriminator. This is composed of an electronic filter which decreases the amplitude of some frequencies relative to others, followed by an AM demodulator. If the filter response changes linearly with frequency, the final analog output will be proportional to the input frequency, as desired. A variant of the Foster-Seeley discriminator called the ratio detector [2] Another method uses two AM demodulators, one tuned to the high end of the band and the other to the low end, and feed the outputs into a difference amp. Using a digital signal processor, as used in software-defined radio.

Q.15 Explain the operation of a 1.pulse averaging discriminator 2.pll discriminator

3.quadrature FM discriminator,appending appropriate diagrams and waveforms.

In an FM signal, the modulation is the deviation of a carrier from its nominal frequency. The conventional method to demodulate this signal is to convert frequency deviation to phase and detect the change of phase. In the quadrature demodulator, the modulated carrier is passed through an LC tank circuit that shifts the signal by 90 at the center frequency. This phase shift is either greater or less than 90 depending on the direction of deviation. A phase detector compares the phaseshifted signal to the original to give the demodulated baseband signal. You use quadrature demodulators not only for frequency modulation, but also with digital modulation schemes such as FSK (frequency shift keying) and GFSK (Gaussian frequency shift keying). FM Quadrature Demodulator Block Diagram The conventional method of FM demodulation for integrated circuits is Bilotti's quadrature demodulator that uses a phase shift network and a phase detector . Figure 1 shows the block diagram

of this quadrature demodulator. The phase detector compares the phase of the IF signal (v1) to v2, the signal generated by passing v1 through a phase shift network. This phase shift network includes an LC tank (L, Rp, and Cp) and a series reactance(Cs). The network gives a frequency-sensitive 90 phase shift at the center frequency. The phase detector discussed here is the bipolar double-balanced multiplier popularized by Bilotti . The output of the multiplier (Io) is filtered, which results in a DC level that changes as the input frequency changes.

Figure 1:

Quadrature demodulator block diagram

Quadrature Demodulator Transfer Function To derive the transfer function of the quadrature demodulator, the phase shift network is first drawn as a small-signal circuit model (Figure 2). The impedance (Zp) of the parallel combination of L, Rp, and Cp is:

Figure 2: Small-signal model of the quadrature phase-shift network The ratio of v2 over v1 is the ratio of impedances Zp(s) over (Zp(s) + 1/sCs). Simplifying this ratio,

The resonant frequency

of this filter is:

The quality factor Q of the phase shift network is Rp/( nL). Next, Equation 2 is used to solve for the transfer function from v1 to v2. The variables n and Qare substituted into Equation 2 and v2/v1 is written in terms of s=j where n:

In Equation 4, is the deviation from the carrier frequency, and 2Q / n is the normalized deviation. Defining:

Equation 4 can be written as:

Writing v2 in terms of v1,

Equation 7 describes the signal at one multiplier input in terms of the signal at the other input. The signal v1 is applied to the first input and is in limiting (a square wave). The signal at the second input (v2) is a linear signal. By integrating over half of the period, you get the average value of the multiplier output current:

For a bipolar differential amplifier, gm is Io/VT where 2Io is the multiplier bias current. Substituting for v2 and gm, where V1 is the peak voltage of the signal v1. Simplifying Equation 9 yields the transfer function for the quadrature demodulator:

In Figure 3, the term a/(1+a) from Equation 10 is plotted versus the normalized frequency deviation (a). This plot is the quadrature

demodulator s-curve. As the frequency of the signal applied to the demodulator becomes more positive than the natural frequency of the phase shift network, the filtered output of the multiplier increases. Likewise, the filtered output decreases as the frequency of the input signal decreases.

Figure 3: Plot of normalized demodulator output vs. normalized frequency deviation Integrated Circuit Implementation Figure 4 shows an integrated circuit implementation of the quadrature demodulator. The input signal vin is supplied from a limiting amplifier and is a square wave of known amplitude. The input signal vin is level shifted, and v1 is applied to transistors Q1 and Q2. The amplitude of v1 is large enough such thatQ1 and Q2 are switched completely on or off during each cycle. Capacitor Cs is typically integrated while Cp, L, and Rp are external components. The component values are chosen such that the amplitude of v2 is less than that ofv1 as given by Equation 7. This causes transistors Q3Q6 to operate as linear devices rather than switches. The output of the multiplier is converted

from a differential current to a single-ended voltage vo. The output is filtered by components Rf and Cf.

Figure 4: Integrated circuit implementation of the quadrature demodulator The authors gratefully acknowledge Mark Randol.