Delay spread in the time domain translates directly into frequency-selective fading in the frequency domain. We assume there are two multipaths having the same amplitude A, as shown in Figure 2. The frequency-selective fading is thus evident in the nulls of the magnitude spectrum as a result of multipath delay. Exercise 2. Radio Propagation 27 References  Lee, W. VT, , pp.
Select Bibliography Lee, W. Lee, W. Rappaport, T. We approach the subject from a system perspective. Namely, we treat the subject in terms of subsystems and block diagrams that carry out specific functions in the overall scheme of moving information from the transmitter to the receiver.
Before we begin, it is important to understand why the wireless personal communications industry is moving from the traditional analog FM technology to digital technologies such as TDMA and CDMA. There are many advantages for migrating to digital, but there are at least four reasons for the recent trend in the industry, as follows. The first is the quality of service. Digital communication systems, as manifested in various technologies, have the potential to offer a higher quality of service.
This is particularly true in harsh radio environments such as the mobile channel. Figure 3. In this simplified digital communication system, a positive pulse which could be designated as a 1 is sent from the transmitter. As the pulse travels to the receiver, it suffers impairment in both amplitude and shape. The shape of the pulse is also distorted, partly due to the lowpass nature of the channel. At point A, the signal has suffered so much degradation that if it were an analog signal, the low signal-to-noise ratio would result in poor signal quality.
However, the digital communication system shown has a receiver that contains a threshold detector. In this case, the cumulative energy is greater than 0, thus the threshold detector makes a decision of 1 and perfectly regenerates the transmitted pulse. In this example, the information is contained in the amplitude of the signal. Other digital communication systems, such as the CDMA system, use phase-shift keying PSK , where the information is contained in the phase of the signal.
The characteristics of PSK are discussed in the modulation section later in this chapter. The second reason is increased capacity. In a voice communication system, the source information is human speech; speech is inherently analog and must be converted coded to a digital form before it Fundamentals of Digital RF Communication 31 can be transmitted via the digital system.
This coding of source information, or source coding, can use speech compression to reduce the number of bits necessary to represent speech. The vocoder therefore uses a smaller bandwidth to transmit speech information. Moreover, the vocoder employed in CDMA is variable rate, which means that during silent period of human speech, the output bit rate of the vocoder is lowered.
The IS vocoder supports four different rates: 9,, 4,, 2,, and 1, bps. The variable-rate feature further reduces the effective bandwidth required to transmit speech. If there is only a fixed amount of total bandwidth, a reduction in the bandwidths of individual voice channels implies an increase in the total number of available voice channels. The third reason is privacy. Privacy is an important issue in any communication system. In the conventional analog FM system, anyone who has an FM scanner can eavesdrop on a conversation.
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A digital communication system provides a ready platform where encryption techniques may be used to safeguard the information transmitted over the air. In a complex system such as CDMA, it would be difficult for anyone to build a receiver and eavesdrop on a conversation, even if the encryption feature is not activated. Complex digital communication systems, especially spread-spectrum ones, are made possible with enabling technologies such as digital signal processing DSP techniques and their implementation on application-specific integrated circuits ASICs.
The cost of these technologies has historically been high enough that they could only be afforded by government and military customers. In recent years, however, the economics of these technologies has made them feasible for use in commercial application on a mass scale. The information source, such as human speech, is first converted into digital form by the source encode function. Then the channel encode function encodes the digital information for the purpose of combating various degrading effects of the channel.
Then the information is arranged by the multiple access function so that more than one user can share the given spectrum. RF waveform that can be transmitted by the transmitter. On the receive side, the bandpass waveform is intercepted by the receiver. The signals are first demodulated from RF to baseband, then the multiple access function separates the different users that are sharing the particular spectrum.
Then, the channel decode function attempts to correct the errors that have been introduced by the channel. The source decode function converts the baseband information back to analog speech. One of the techniques used in wireline applications is pulse code modulation PCM , where the analog voice is converted into a Kbps bit stream. This is the reason why computer modems can be used over telephones; the information contained in the waveform generated by a transmitting modem can be reliably received by the receiving modem on the other end, and the Fundamentals of Digital RF Communication 33 reason is that PCM attempts to replicate the waveform regardless of whether or not the information contained in the waveform is human speech or modulated pitches generated by a modem.
PCM is not feasible in wireless applications because there is a limited bandwidth available. Transmitting 64 Kbps of information over the air demands more bandwidth than can be afforded by most service providers. Therefore, alternative source coding techniques are needed to represent source information human speech, in this case using less bandwidth.
A vocoder offers an attractive solution. It exploits the characteristics of human speech and uses fewer bits to represent and replicate human sounds. See Figure 3. The temporal and frequency characteristics of human sound are exploited by vocoders for speech coding. The human voice is made up of a combination of voiced and unvoiced sounds. These sounds have essentially a periodic rate with a fundamental Speech or music or modem feed PCM 64 Kbps Wireline Speech only Vocoder 9. This fundamental frequency is also known as pitch.
These sounds are more like acoustic noise created by a closure and sudden release of vocal tract. Although human voice is time varying, its spectrum is typically stationary over a period between 20 and 40 ms. This is the reason why most vocoders produce frames that have a duration on this order. For example, the IS vocoder produces frames that are 20 ms in duration. That is, the filter response varies with time.
This is done by periodically updating the coefficients of the filter. This filter is typically all-pole because an all-pole filter requires less computational power than a filter with both poles and zeros. Equation 3. This type of coding technique is sometimes called analysis-synthesis coding. The excitation parameters and filter coefficients are the outputs of the speech encoder. In the IS CDMA system, these parameters and coefficients are the information that is communicated between the transmitter and receiver.
The voice decoder at the receiving end uses these parameters and coefficients to construct the excitation source and synthesis filter. Linear-predictive coding LPC is widely used to estimate filter coefficients. A feedback loop in the encoder is used to compare actual voice and replicated voice.
The difference between actual voice and replicated voice is the error. LPC is set up to generate filter coefficients such that this error is minimized. These filter coefficients, along with excitation parameters, are then used by the decoder for speech synthesis. Specifically, the CELP decoder uses a codebook to generate excitation inputs to the synthesis filter. This is done to improve performance of the communication system by enabling the signal to better withstand the effects of channel impairments, such as noise and fading.
The cost of this goal is more bandwidth, or more redundant bits that the system has to transmit . In this section, we deal specifically with error-correcting codes, which when applied to channel coding improve the error performance of the system. The purpose is to add extra bits to the information bits so that errors may be found and corrected at the receiver.
In other words, a sequence of bits is represented by a longer sequence of bits with enough redundancy to protect the data . For example, the simplest error-correcting code is to repeat the information bits. Suppose I have a bit I wish to send and error protect. This way, I will improve the chance that the receiver correctly receives a 1 in case any one of the transmitted bits is flipped to 0 during the transmission process.
In this case, the receiver will use majority decoding. Namely, the receiver will only decide a 1 if a majority of the three bits are received as 1s. This code is known as a 3, 1 code. A code is sometimes described by its rate. Block codes, as the name implies, code an information sequence one block at a time. Convolutional codes, on the other hand, have a memory property.
The memory depends on the constraint length K of the convolutional code. A linear block code can be characterized by the n, k notation, and for a given code, the encoder transforms a block of k information bits into a longer block of n code bits . The code bits are only a function of the current block of information bits. For example, we can define a 7, 4 linear block code where a block of seven code bits is used to represent a block of four information bits. Table 3. This simple 7, 4 linear block code is also known as the 7, 4 Hamming code, and the redundancy bits are also known as the parity bits.
It is intuitive that the extra redundancy bits improve the error performance of the system. To quantify this performance, we introduce the concept of Hamming distance. The Hamming distance between any two code words is the number of places that the two code words differ. For example, the Hamming distance between 1, 1, 1, 1, 1, 1, 1 and 1, 1, 1, 0, 1, 0, 0 is 3.
Minimum distance turns out to be a critical parameter that Table 3. If t errors occur during the transmission of a code word, and if the Hamming distance between the received word and every other code word is larger than t, then the decoder will properly correct the errors if it assumes that the closest code word to the received word was actually transmitted .
On the other hand, equation 3. As mentioned above, in order to decode a received code word, the decoder assumes that the closest code word to the received code word was actually transmitted. For example, suppose that a received code word is 0, 0, 0, 1, 1, 1, 1 ; since this received code word is not one of the specified code words in the 7, 4 Hamming code, an error or errors must have occurred. Assuming that the closest code word to the received code word was actually transmitted, the decoder decides that the code word 0, 0, 0, 1, 0, 1, 1 was actually sent by the transmitter.
In actuality, a digital logic circuit is used to implement the decoder. For CRC, the information bits are treated as one long binary number. This number is divided by a unique prime number that is also binary, and the remainder is appended to the information bits as redundancy bits. When the frame is received, the receiver performs the same division using the same prime divisor and compares the calculated remainder with the remainder received in the frame .
For example, the 7, 4 Hamming code discussed in Section 3.
The polynomial g x is a prime polynomial. Suppose the message 1, 0, 1, 0 needs to be encoded using the 7, 4 Hamming code. As we can see in the 7, 4 Hamming code in Table 3. See  for a good discussion on cyclic redundancy codes. References  and  give a good discussion on cyclic codes in general. In the IS CDMA system, when the vocoder is operating at full rate, each ms frame contains bits, which are made up of information bits, 12 frame quality bits, and 8 encoder tail bits. The 8 encoder tail bits are all set to 0.
The 12 frame quality indicator bits are the redundancy bits that are a function of the information bits in the frame. The convolutional codes, on the other hand, do have memory. For convolutional codes, the encoded bits are functions of information bits and functions of the constraint length. Specifically, every encoded bit at the output of the convolutional encoder is a linear combination of some previous information bits. Initially, all the registers are initialized to zero. As the information message bits mi are clocked in from the left, bits are tapped off different stages of the delay line and summed in the modulo-2 adder.
The summation is the output of the convolutional encoder. A commutator switch toggles through both output points for every input clock cycle; hence, the output rate is mi y 'i Modulo 2 adder y ''i Modulo 2 adder Figure 3.
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The generator function for the two output bits y 'i and y ''i shown in Figure 3. Since the mobile has a limited transmit power, the reverse link can sometimes be the limiting link. In this case, three bits are generated for every input bit, and the output rate is effectively three times the input rate. See  for a good discussion of convolutional decoding. Fundamentals of Digital RF Communication 43 3. The error-correcting codes are designed to combat errors resulting from fades and, at the same time, keep the signal power at a reasonable level. Most error-correcting codes perform well in correcting random errors.
However, during periods of deep fades, long streams of successive or burst errors may render the error-correcting function useless. Interleaving is a technique for randomizing the bits in a message stream so that burst errors introduced by the channel can be converted to random errors. In Figure 3. One way to interleave the message is to load it into a matrix of four rows and ten columns. We truncate the message into four parts and load them into the four rows.
Then we read the message out from the top, column by column. The resulting randomized message is sent through the channel. The channel introduces several burst errors into the message. As a result, the underlined alphabets are received in error. At the receiving end, a deinterleaver reconstructs the message using the same matrix, except in this case the deinterleaver loads the received message into columns first, then reads the message out from the rows.
As we can see, the burst errors are indeed converted to scattered random errors. In this case, the interleave depth is The interleaver for the reverse traffic channel uses a matrix of 32 rows by 18 columns at full rate. The interleaver for the forward traffic channel uses a matrix of 24 rows by 16 columns at full rate. Multiple access refers to the sharing of a common resource in order to allow simultaneous communications by multiple users, and this common resource is the RF spectrum. In the traditional FDMA scheme, each individual user is assigned a particular frequency band in which transmission can be carried out see Figure 3.
A portion of the frequency spectrum is divided into different channels. This way, multiple users can simultaneously share the frequency spectrum. Each of the codes are orthogonal to one another, and channelization of simultaneous users is achieved by the use of this set of orthogonal codes. All the spread wideband signals of different users are added together to form a composite signal, and the composite signal is transmitted over the air in the same frequency band.
The receiver sifts the desired user out of the composite signal by correlating the composite signal with the original code. All other users with codes that do not match the code of the desired user are rejected. The differences range from the modulation scheme to error control methods. In addition, each link uses different codes to channelize individual users. The forward link uses Walsh codes, while the reverse link uses pseudorandom noice PN codes for channelization. In order to avoid mutual interference on the forward link, Walsh codes are used to separate individual users while they simultaneously occupy the same RF band.
Walsh codes as used in IS are a set of 64 binary orthogonal sequences. These sequences are orthogonal to each other, and they are generated by using the Hadamard matrix. The three conditions are 1. The scaled dot product of each code should equal to 1. The readers can easily verify that all of the above sequences except w 0 satisfy the conditions. For example, 8 orthogonal sequences, each of length 8, can be obtained by generating H8; 16 orthogonal sequences, each of length 16, can be obtained by generating H The IS forward link uses a set of 64 orthogonal Walsh sequences, thus the physical limitation on the number of channels on the forward link is 63 because in an IS system, w 0 is not used to transmit any baseband information.
Suppose that there are three different users, and each user wishes to send a separate message. Note that the chip rate of the Walsh code is four times the bit rate of the message, contributing to a processing gain of 4. If there are negligible errors during the transmission process, the receiver intercepts C t. However, the ability to channelize depends heavily on the orthogonality of the code sequences during all stages of the transmission. In practice, the IS CDMA system uses a pilot channel and a sync channel to synchronize the forward link and to ensure that the link is coherent. The mobile stations transmit at will, and no attempt is made to synchronize their transmissions.
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Thus Walsh codes cannot be used on the reverse link. The incoherent nature of the reverse link calls for the use of another class of codes, PN codes, for channelization. One such example a three-stage register is shown in Figure 3. Binary bits are shifted through the different stages of the register. The output of the last stage and the output of one intermediate stage are combined and fed as input to the first stage. The register starts with an initial sequence of bits, or initial state, stored in its stages.
Then the register is clocked, and bits are moved through the stages.
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This way, the register continues to generate output bits and feed input bits to its first stage. The output bits of the last stage form the PN code. We now demonstrate the code generation using the register shown in Figure 3. An initial state of [1, 0, 1] is used for the register. The output of stage 3 is the output of the register. After clocking the bits through the register, we obtain the results summarized in Table 3. Thus, the effective length of the periodic PN code generated is 7.
The PN code structure is determined by the feedback logic i. Since the maximal PN code length is always an odd number see 3. Suppose the same three users wish to send three separate messages. Each message is spread by its assigned PN code. Note that the chip rate of the PN code is seven times the bit rate of the message, contributing to a processing gain of 7. We proceed to calculate the autocorrelation of the PN sequence p0. The resulting R p 0 i for each shift i is shown on the right side of the table. The autocorrelation property shown in Table 3.
A high correlation occurs only when the codes are aligned i.
In practice, the receiver possesses an original copy of the PN code i. When the autocorrelation reaches a maximum, then the two codes are in-phase and have a time shift of zero. This acquisition scheme can also be used when the spreading code length is equal to the data bit period. The long code is so called because its length is literally very long. We saw in Section 3. However, the forward link also uses the PN code. Each base station is assigned a unique PN code that is superimposed on top of the Walsh code. This is done to provide isolation 58 CDMA RF System Engineering among the different base stations or sectors ; the isolation is necessary because each base station uses the same 64 Walsh code set.
It is so called because its length is relatively short. The modulated signal is then transmitted through space in the form of a propagating electromagnetic EM field. There are two answers to this question. First, the government regulatory agency i. Thus, not everyone can transmit at the baseband frequency.
Second, in order to transmit at baseband, which is at a much lower frequency, the required antenna size would be enormous in order to allow an efficient coupling between the transmitter and free space. For example, in order to efficiently couple power to free space, the antenna size needs to be at least on the order of the wavelength. If one wishes to transmit a baseband signal at 9. There is a difference between analog and digital modulation techniques. Readers may be familiar with analog modulation schemes such as amplitude modulation AM and frequency modulation FM.
In analog modulation, information is contained in the continuous-waveform shape of the signal. Digital modulation schemes, on the other hand, are used to transmit discrete units of information called symbols, and the information may be contained in the amplitude e. The concept is simple. The BPSK modulator is quite simple to implement. The modulator itself is no more than a multiplier.
The input to the modulator consists of the data symbols. The output of the multiplier is the corresponding modulated signal. The multiplied result is then integrated over one bit interval T. Furthermore, it is assumed that the demodulator is coherent i. The noise terms i. If the noise power is large i. The signal-space representation is nothing more than another representation of the signals. The representation depicts a signal in its in-phase and quadrature components. Every real-valued signal can be de-composed into an I and a Q component, and the signal-space representation effectively draws the signal in a space defined by the I and the Q axes.
The in-phase component is calculated by multiplying that signal by a cosinusoid and then integrating over the bit period. Note that s is a constant in the IQ representation, and n is effectively a Gaussian random variable; thus, y is also Gaussian distributed with a mean of s. This probability can be evaluated by integrating the Gaussian probability density function over the error area. In this section, we examine another modulation scheme that can transmit two bits of information per symbol period.
Quadrature phase-shift keying QPSK makes use of the quadrature component in addition to the inphase component. The mapping shown in Table 3. In order to transmit four different symbols, the QPSK transmitter needs to be able to send four different waveforms: s 0 t , s1 t , s 2 t , and s 3 t. Each signal corresponds to one of the four symbols; that is, Table 3. The transmitter changes the phase of the cosine waveform, depending on which symbol is to be transmitted. The phase can change to any one of the four states: 45, , , and degrees.
The data bits are fed into a demultiplexer DEMUX , where the bit stream is separated into an even bit stream and an odd bit stream. The even bit stream is multiplied by the in-phase carrier, and the odd bit stream is multiplied by the quadrature carrier. The output of the multipliers is combined in the adder to form the QPSK signal. The multiplied results are integrated and fed into the decision threshold. Furthermore, it is assumed that the demodulator is coherent, which means that the phases of the reference signals perfectly match the phases of the modulating carriers. The noise variance i.
An error would also occur if symbol 0 is pushed into decision region C; however, this case is considered less likely because the noise power needed is greater than that of the two previously mentioned cases. Therefore, we can express 3. By symmetry, the probability that symbol 0 would drift into decision D is also P. The probability that symbol 0 would be in error is thus equal to the sum of the two probabilities since the occurrence of either of the two events would trigger an error.
To state this differently, only one of the two bits changes when a symbol transitions to its nearest neighbor. Therefore, whenever a symbol error occurs, only one of the two bits contained in that symbol is also in error. This is done to avoid a degree phase transition that occurs in conventional QPSK systems. For example, when symbol 0 transitions to symbol 3, the signal goes through a degree phase transition through the origin. In the time domain, the signal envelope collapses and momentarily reaches zero. This zero crossing demands a lot of dynamic range from the power amplifier.
Thus, OQPSK is used on the reverse link where the power amplifier of the mobile is limited both in size and in performance. The extra delay of half a bit in the Q path ensures that there will be no direct transition between symbols 0 and 2 and between symbols 1 and 3, and thus no zero crossing. It is also important to note that the error performance i. In addition, the mobile communications environment is characterized by fading channels rather than the AWGN channel. References  Sklar, B. IEEE Conf. Acoustic, Speech, and Signal Processing, Costello, Jr. Select Bibliography Gagliardi, R.
Haykin, S. Jayant, N. Markel, J. Gray, Jr. Papoulis, A. Schafer, R. Simon, M. Hinedi, and W. Viterbi, A. Although all users are transmitting in the same RF band, individual users are separated from each other via the use of orthogonal codes. The rate of the final spread signal is 1.
There can be many 1. To a large degree, the performance of a CDMA system is interference-limited. This means that the capacity and quality of the system are limited by the amount of interference power present in the band. Capacity is defined as the total number of simultaneous users the system can support, and quality is defined as the perceived condition of a radio link assigned to a particular user; this perceived link quality is directly related to the probability of bit error, or bit error rate BER.
This chapter presents those characteristics of a CDMA system that need to be optimized in order to reduce interference and increase quality. The actual capacity of a CDMA cell depends on many different factors, such as receiver demodulation, power-control accuracy, and actual interference power introduced by other users in the same cell and in neighboring cells. Notice that 4. We can further manipulate 4. Here, we consider the reverse-link capacity since in CDMA this is often the limiting link in terms of capacity. Reverse link is the mobile to base station link.
We assume that the system possesses perfect power control, which means that the transmitted powers of all mobile users are actively controlled such that at the base station receiver, the received powers from all mobile users are equal. This is so because the total interference power in the band is equal to the sum of powers from individual users. Figure 4. Note that 4. We proceed to substitute 4.
Therefore, if there are seven users occupying the band, and each user is power-controlled to the same power level, then the SNR experienced by any one user is 1 6. Although these other users from other cells are power-controlled by their respective home cells, the signal powers from these other users constitute interference to cell A. Therefore, cell A is said to be loaded by users from other cells. Equation 4. In the example shown in Figure 4. Instead of having an omnidirectional antenna, which has an antenna pattern over degrees, cell A can be sectorized to three sectors so that each sector is only receiving signals over degrees see Figure 4.
In effect, a sectorized antenna rejects interference from users that are not within its antenna pattern. If the cell is sectorized to six sectors, then the loading effect is decreased by a factor of approximately 6. The integrals in 4. For example, if the user is not speaking during part of the conversation, the output rate of the vocoder is lowered to prevent power from being transmitted unnecessarily.
The effect of this variable-rate vocoding is the reduction of overall transmitted power and hence interference. By employing variable-rate vocoding, the system reduces the total interference power by this voice activity factor. Thus, 4. Note that the effect of voice activity is to reduce the denominator, or the interference portion of the equation. Solving 4. Capacity, or number of simultaneous users M, is directly proportional to the processing gain of the system. Principles of Code Division Multiple Access 83 2. Capacity can be increased if one can decrease the amount of loading from users in adjacent cells.
Spatial filtering, such as sectorization, increases system capacity. For example, a six-sector cell would have more capacity than a threesector cell. Power control is essential to the smooth operation of a CDMA system. Because all users share the same RF band through the use of PN codes, each user looks like random noise to other users. The power of each individual user, therefore, must be carefully controlled so that no one user is unnecessarily interfering with others who are sharing the same band.
To illustrate how power control is essential in CDMA, we consider a single cell that has two hypothetical users see Figure 4. We again examine the reverse-link case since this link is often the limiting link in CDMA. User 2 is much closer to the base station than user 1. If there is no power control, both pt User 1 pr,1 pr,2 pt User 2 Figure 4. Each user is transmitting to the base station a fixed amount of power pt. If we assume that the difference in distance is such that pr ,2 is 10 times more than pr ,1 , then user 1 would be at a great disadvantage.
This inequity is known as the classic near-far problem in a spread-spectrum multiple access system. The system at this point is said to have reached its capacity. Power control is implemented to overcome the near-far problem and to maximize capacity. User 2 has a much higher SNR than user 1 does. Principles of Code Division Multiple Access 85 to one other. In the cell, if the transmit power of each user is controlled such that the received power of each user at the base station is equal to pr , then a lot more users can be accommodated by the system.
The capacity is maximized with the use of power control see Figure 4. Before the mobile establishes contact with the base station, the mobile cannot be power-controlled by the base station. Thus, the natural question is when the mobile first attempts to access the base station, what power level should the mobile use to transmit its request? At this point, the base station has not yet made contact with the mobile user, and the base station has no idea as to the location of the mobile user.
There are two options: the first option is that the mobile can attempt to access the base station with a high transmit power. However, the disadvantage of a high initial transmit power is that such high power represents interference to pt,3 User 3 pr User 1 pr pr pt,1 pr pt,M pt,2 User 2 User M Figure 4. As is typical for mobile communication systems standardization, sufficient RF performance has been assumed and most efforts have been put to digital baseband issues.
This is especially true for the mobile phone transceivers, the RF part of which is although its baseband part is much more complex in terms of number of devices still the bottleneck of the entire system. Meanwhile, in the RF concept engineering of today's commercial products, an accurate prediction of the needed RF performance by using RF system simulation is indispensable.
The paper gives an insight of how to derive receiver requirements for third-generation mobiles in terms recognizable by microwave designers, reports on the system simulation-based design, and performance of silicon-based RF integrated circuits for mobile terminal use, and discusses some future technologies and techniques and their possible impact on portable wireless devices.
Article :. Connolly noted that the certificate is designed for people with an engineering or technical education, however, knowledge of telecommunications engineering is not necessary. He said several types of individuals have been identified as the program's target audience, including: -Electrical engineers who have been out of school for a number of years and want to transition into the booming telecommunications industry; -Newly graduated engineers looking to have the applied communication engineering skills to make them attractive as potential employees of high-tech telecom companies; -Under -graduate engineering students, enrolled in the University's Concurrent Enrollment Program, so that on graduation these students increase their marketability to high-tech telecom companies; and -System engineers, telecommunication network planners and other engineers who wish to gain more in-depth knowledge of digital communication system, with an emphasis on CDMA technology.
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