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CHAPTER 1 INTRODUCTION
OFDM is of great interest by researchers and research laboratories all over the world. It has already been accepted for the new wireless local area network standards IEEE 802.11a, High Performance LAN type 2 (HIPERLAN/2) and Mobile Multimedia Access Communication (MMAC) Systems. Also, it is expected to be used for wireless broadband multimedia communications.Data rate is really what broadband is about. The new standard specify bit rates of up to 54 Mbps. Such high rate imposes large bandwidth, thus pushing carriers for values higher than UHF band. For instance, IEEE802.11a has frequencies allocated in the 5- and 17- GHz bands. This project is oriented to the application of OFDM to the standard IEEE 802.11a, following the parameters established for that case. OFDM can be seen as either a modulation technique or a multiplexing technique. One of the main reasons to use OFDM is to increase the robustness against frequency selective fading or narrowband interference. In a single carrier system, a single fade or interferer can cause the entire link to fail, but in a multicarrier system, only a small percentage of the subcarriers will be affected. Error correction coding can then be used to correct for the few erroneous subcarriers. The concept of using parallel data transmission and frequency division multiplexing was published in the mid-1960s. In a classical parallel data system, the total signal frequency band is divided into N nonoverlapping frequency subchannels. Each subchannel is modulated with a separate symbol and then the N subchannels are frequency-multiplexed.It seems good to avoid spectral overlap of channels to eliminate interchannel interference. However, this leads to inefficient use of the available spectrum.To cope with the inefficiency, the ideas proposed from the mid-1960s were to use parallel data and FDM with overlapping subchannels, in which, each carrying a signaling rate b is spaced b apart in frequency to avoid the use of high-speed equalization and to combat impulsive noise and multipath distortion, as well as to fully use the available bandwidth.
Figure 1.1: Concept of OFDM signal: orthogonal multicarrier technique versus conventional multicarrier technique
Pictorially it can be represented as shown in the figure (1) in the next page. The figure shows the difference between the conventional non-overlapping multicarrier technique and overlapping multicarrier modulation technique. As shown in figure 1, by using the overlapping multicarrier modulation technique, we save almost 50% of bandwidth. To
realize the overlapping multicarrier technique, however we need to reduce crosstalk between subcarriers, which means that we want orthogonality between the different modulated carriers. The orthogonality of the carriers means that each carrier has an integer number of cycles over a symbol period. Due to this, the spectrum of each carrier has a null at the center frequency of each of the other carriers in the system. This results in no interference between the carriers, allowing then to be spaced as close as theoretically possible. This overcomes the problem of overhead carrier spacing required in FDMA. Each carrier in an OFDM signal has a very narrow bandwidth (i.e.1kHz), thus the resulting symbol rate is low. This results in the signal having a high tolerance to multipath delay spread, as the delay spread must be very long to cause significant inter-symbol interference (e.g. > 500 sec).
The carriers of an OFDM are sinusoids that meet this requirement because each one is sa multiple of frequency. Each one has an integer number of cycles in the fundamental period.
1.3 Modulation
Modulation is the process of modifying some properties of the high frequency carrier signal in accordance with the baseband signal. Binary data from the memory device or from a digital processing stream is used as the modulating signal. The following steps may be carried out in order to apply modulation to the carriers in OFDM: Combine the binary data into symbols according to the number of bits/ symbols selected. Convert the serial symbols stream into parallel segments according to the number of carrier and form the carrier symbol sequence. Apply differential coding to each carrier symbol sequence. Convert each symbol into complex phase representation. Assign each carrier sequence to the appropriate IFFT bin, including complex conjugate. Take IFFT of the result.
of 1/NT [4] The peak (for each carrier) is at the carrier frequency k/NT. Therefore, each carrier frequency is located at the nulls for all the other carriers. This means that none of the carriers will interfere with each other during transmission, although their spectrums overlap. The ability to space carriers so closely together is very bandwidth efficient. In the process of transmission and reception it is essentially required to linearly amplify the signals. This is a sort of disadvantage of the OFDM system.
1.5 Demodulation
This process is the juts reverse of the modulation process. It is carried out on the receiver side of the system and is done in the frequency domain. The following steps may be taken to demodulate the OFDM signal: Partition the input stream into vectors representing each symbol period. Take the FFT of each symbol period vector. Extract the carrier FFT bins and calculate the phase of each. Calculate the phase difference, from one symbol period to the next, for each carrier. Decode each phase into binary data. Sort the data into appropriate order.
Using adequate channel coding and interleaving one can recover symbols lost due to frequency selectivity of the channel. Channel equalization becomes simpler than single carrier system by using adaptive equalization techniques. In conjunction with differential modulation there is no need to implement a channel estimator. It is less sensitive to sample timing offset than the single carrier system. Provides good protection against co-channel interference and impulsive parasitic noise.Though the OFDM scheme has numerous advantages, there are still some drawbacks in this scheme. They are indicated as below:
The OFDM signal has a high Peak to Average Power Ratio (PAPR) It is more sensitive to carrier frequency offset and drift than the single carrier systems dueto leakage in the DFT. Phase noise and Image Rejection are also a problem in OFDM .
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3.1 Definition
Signal-to-noise ratio is defined as the power ratio between a signal (meaningful information) and the background noise (unwanted signal):
where P is average power. Both signal and noise power must be measured at the same or equivalent points in a system, and within the same system bandwidth. If the signal and the noise are measured across the sameimpedance, then the SNR can be obtained by calculating the square of the amplitude ratio:
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where A is root mean square (RMS) amplitude (for example, RMS voltage). Because many signals have a very wide dynamic range, SNRs are often expressed using the logarithmic decibel scale. In decibels, the SNR is defined as
The concepts of signal-to-noise ratio and dynamic range are closely related. Dynamic range measures the ratio between the strongest un-distorted signal on a channel and the minimum discernable signal, which for most purposes is the noise level. SNR measures the ratio between an arbitrary signal level (not necessarily the most powerful signal possible) and noise. Measuring signal-to-noise ratios requires the selection of a representative orreference signal. In audio engineering, the reference signal is usually a sine wave at a standardized nominal or alignment level, such as 1 kHz at +4 dBu (1.228 VRMS). SNR is usually taken to indicate an average signal-to-noise ratio, as it is possible that (near) instantaneous signal-to-noise ratios will be considerably different. The concept can be understood as normalizing the noise level to 1 (0 dB) and measuring how far the signal 'stands out'.
so
that usually we don't include that resistance term while measuring power or energy of a signal. This usually causes some confusions among readers but the resistance term is
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not significant for operations performed in signal processing. Most of cases the power of a signal would be
where 'A' is the amplitude of the ac signal. In some places people just use
where
noise, or an estimate thereof. Notice that such an alternative definition is only useful for variables that are always non-negative (such as photon counts and luminance). Thus it is commonly used in image processing, where the SNR of an image is usually calculated as the ratio of the mean pixel value to the standard deviation of the pixel values over a given neighborhood. Sometimes SNR is defined as the square of the alternative definition above. The Rose criterion (named after Albert Rose) states that an SNR of at least 5 is needed to be able to distinguish image features at 100% certainty. An SNR less than 5 means less than 100% certainty in identifying image details. Yet another alternative, very specific and distinct definition of SNR is employed to characterize sensitivity of imaging systems; see signal to noise ratio (imaging). Related measures are the "contrast ratio" and the "contrast-to-noise ratio".
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where W is the bandwidth and ka is modulation index. Output signal-to-noise ratio (of AM receiver) is given by
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depending on what is measured and of the sensitivity of the device. It is often possible to reduce the noise by controlling the environment. Otherwise, when the characteristics of the noise are known and are different from the signals, it is possible to filter it or to process the signal. For example, it is sometimes possible to use a lock-in amplifier to modulate and confine the signal within a very narrow bandwidth and then filter the detected signal to the narrow band where it resides, thereby eliminating most of the broadband noise. When the signal is constant or periodic and the noise is random, it is possible to enhance the SNR by averaging the measurement. In this case the noise goes down as the square root of the number of averaged samples.
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that further decrease the SNR compared to the theoretical maximum from the idealized quantization noise, including the intentional addition of dither. Although noise levels in a digital system can be expressed using SNR, it is more common to use Eb/No, the energy per bit per noise power spectral density. The modulation error ratio (MER) is a measure of the SNR in a digitally modulated signal.
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to noise, interference, distortion or bit synchronization errors. The bit error rate or bit error ratio (BER) is the number of bit errors divided by the total number of transferred bits during a studied time interval. BER is a unitless performance measure, often expressed as a percentage. The bit error probability pe is the expectation value of the BER. The BER can be considered as an approximate estimate of the bit error probability. This estimate is accurate for a long time interval and a high number of bit errors. Example: As an example, assume this transmitted bit sequence: 0 1 1 0 0 0 1 0 1 1, and the following received bit sequence: 0 0 1 0 1 0 1 0 0 1, The number of bit errors (the underlined bits) is in this case 3. The BER is 3 incorrect bits divided by 10 transferred bits, resulting in a BER of 0.3 or 30%.
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or line
coding scheme,
and
by
applying channel
codingschemes
such
as
redundant forward error correction codes. The transmission BER is the number of detected bits that are incorrect before error correction, divided by the total number of transferred bits (including redundant error codes). The information BER, approximately equal to the decoding error probability, is the number of decoded bits that remain incorrect after the error correction, divided by the total number of decoded bits (the useful information). Normally the transmission BER is larger than the information BER. The information BER is affected by the strength of the forward error correction code.
Binary symmetric channel (used in analysis of decoding error probability in case of non-bursty bit errors on the transmission channel)
A worst case scenario is a completely random channel, where noise totally dominates over the useful signal. This results in a transmission BER of 50% (provided that a Bernoulli binary data source and a binary symmetrical channel are assumed, see below).
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Figure 4.1: Bit-error rate curves for BPSK, QPSK, 8-PSK and 16-PSK, AWGN channel.
In a noisy channel, the BER is often expressed as a function of the normalized carrierto-noise ratio measure denoted Eb/N0, (energy per bit to noise power spectral density ratio), or Es/N0 (energy per modulation symbol to noise spectral density). For example, in the case of QPSK modulation and AWGN channel, the BER as function of the Eb/N0 is given by: .
People usually plot the BER curves to describe the functionality of a digital communication system. In optical communication, BER(dB) vs. Received
Power(dBm) is usually used; while in wireless communication, BER(dB) vs. SNR(dB) is used. Measuring the bit error ratio helps people choose the appropriate forward error correction codes. Since most such codes correct only bit-flips, but not bit-insertions or bit-deletions, the Hamming distance metric is the appropriate way to measure the number of bit errors. Many FEC coders also continuously measure the current BER. A more general way of measuring the number of bit errors is the Levenshtein distance. The Levenshtein distance measurement is more appropriate for measuring raw channel performance before frame synchronization, and when using error correction codes designed to correct bit-insertions and bit-deletions, such as Marker Codes and Watermark Codes.
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Pattern Generator, which transmits a defined test pattern to the DUT or test system.
Error detector connected to the DUT or test system, to count the errors generated by the DUT or test system.
Clock signal generator to synchronize the pattern generator and the error detector.
Electrical-optical converter and optical-electrical converter for testing optical communication signals.
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synchronization. The details of each module in the QAM system will be discussed in the following section. There is an additional difference between the QAM system and the CPM system, namely, the CPM system does not have a baseband filter nor does it has a synchronizer.
s
Figure 5.2: QAM transceiver system
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The block diagram of a QAM transmitter is shown in Figure. The information sequence u is encoded into a code sequence c by the channel encoder. The coded bits c are then interleaved at the bit level to produce the sequence v. The interleaved bits are then modulated to produce 16-QAM or 32-QAM symbols. Physical hops are formed by adding pilots in the middle of a block of data symbols. Finally, the hops are filtered with a baseband filter to meet the requirements of the spectrum mask and amplified by a nonlinear high power amplifier.
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same number of QAM symbols and to insure that 8 or 16 physical hops make up a single codeword. Code bits are grouped into binary 4-tuples or 5-tuples and are modulated using 16-QAM or 32-QAM methods. The outputs of the modulator are separated into 8 or 16 hops and a pilot sequence is inserted into the middle of each hop.3 The pilots will be used in synchronization and channel estimation.
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A block diagram of the receiver is shown in Figure(10). The received signals are first sampled at the sampling rate 4fsymbol. The synchronizer then filters the discrete time signal r(nTs) with a bank of matched filters to recover the symbol timing. The channel estimator estimates the channel impulse response H(n) using the pilot sequence. The pilots are also used to estimate the carrier frequency. The matched filter output y(kTsym) is equalized and demapped into log-likelihood ratios for each encoded bit. Finally, the turbo decoder accepts the log-likelihood ratios and executes the iterative decoding algorithm to produce an estimate of the data bits.
5.7 Synchronizer
Symbol timing synchronization is designed to recover data from a digitally modulated waveform. In the system under evaluation, a polyphase filterbank is used for symbol timing synchronization. The synchronizer filters the discrete time signal r(nTs) with a bank of matched filters to recover the symbol timing.
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Binary signaling is used, i.e., M = 2. It uses rectangular pulse shaping, and the length of the shaping function is larger than 1.So the schemes used in the simulation are partial response CPM. A small modulation index is used to improve the spectral efficiency.
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they can use power efficient nonlinear amplifiers. The complexity of system implementation is another important consideration. With more complicated systems, come more power consumption and longer processing delay. In the design of a communication system, there is always a tradeoff between power and spectral efficiency. Different modulation schemes have different characteristics that call for different levels of the tradeoff between power and spectral efficiency. When using a linear power amplifier, QAM is always superior to CPM. However, QAM will have performance degradation if a nonlinear amplifier is used; in contrast, a CPM system will not be affected as much by amplifier nonlinearities. As a result, we have to reduce the spectral efficiency to compensate for the loss in power efficiency for a QAM system; in contrast, the energy efficiency of a CPM system can be maximized while keeping the spectrum efficiency the same.
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Specifying the Encoder -To define the convolutional encoder, use the Trellis structure parameter. This parameter is a MATLAB structure whose format is described in the Trellis Description of a Convolutional Encoder section of the Communications Toolbox documentation. You can use this parameter field in two ways:
If you have a variable in the MATLAB workspace that contains the trellis structure, enter its name in the Trellis structure parameter. This way is preferable because it causes Simulink to spend less time updating the diagram at the beginning of each simulation, compared to the usage described next.
If you want to specify the encoder using its constraint length, generator polynomials, and possibly feedback connection polynomials, use a poly2trellis command in the Trellis structure parameter. For example, to use an encoder with a constraint length of 7, code generator polynomials of 171 and 133 (in octal numbers), and a feedback connection of 171 (in octal), set the Trellis structure parameter to poly2trellis(7,[171 133],171).
Number of rows :
The number of rows in the matrix that the block uses for its computations.
Number of columns :
The number of columns in the matrix that the block uses for its computations.
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M-ary number-:The number of points in the signal constellation. It must have the form 2K for some positive integer K. Input type: Indicates whether the input consists of integers or groups of bits. Constellation ordering: Determines how the block maps each symbol to a group of output bits or integer. Selecting User-defined displays the field Constellation mapping, which allows for user-specified mapping.
Constellation mapping: This parameter is a row or column vector of size M and must have unique integer values in the range [0, M-1]. The values must be of data type double. The first element of this vector corresponds to the topleftmost point of the constellation, with subsequent elements running down column-wise, from left to right. The last element corresponds to the bottomrightmost point. This field appears when User-defined is selected in the dropdown list Constellation ordering.
Normalization method : Determines how the block scales the signal constellation. Choices are Min. distance between symbols, Average Power, and Peak Power.
Minimum distance : The distance between two nearest constellation points. This field appears only when Normalization method is set to Min. distance between symbols.
Average power, referenced to 1 ohm (watts) : The average power of the symbols in the constellation, referenced to 1 ohm. This field appears only when Normalization method is set to Average Power.
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Peak power, referenced to 1 ohm (watts) : The maximum power of the symbols in the constellation, referenced to 1 ohm. This field appears only when Normalization method is set to Peak Power.
Phase offset (rad) : The rotation of the signal constellation, in radians. Output data type : The output data type can be set to double, single, Fixedpoint, User-defined, or Inherit via back propagation. Setting this parameter to Fixed-point or User-defined enables fields in which you can further specify details. Setting this parameter to Inherit via back propagation, sets the output data type and scaling to match the following block.
Output word length : Specify the word length, in bits, of the fixed-point output data type. This parameter is only visible when you select Fixed-point for the Output data type parameter.
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7.6 NORMALIZE
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Es = Signal energy (Joules) Eb = Bit energy (Joules) N0 = Noise power spectral density (Watts/Hz) Tsym is the Symbol period parameter of the block in Es/No mode k is the number of information bits per input symbol Tsamp is the inherited sample time of the block, in seconds
For real signal inputs, the AWGN Channel block relates Es/N0 and SNR according to the following equation: Es/N0 = 0.5 (Tsym/Tsamp) SNR
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Note that the equation for the real case differs from the corresponding equation for the complex case by a factor of 2. This is so because the block uses a noise power spectral density of N0/2 Watts/Hz for real input signals, versus N0 Watts/Hz for complex signals.
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The Rectangular QAM Demodulator Baseband block demodulates a signal that was modulated using quadrature amplitude modulation with a constellation on a rectangular lattice. The signal constellation has M points, where M is the M-ary number parameter. M must have the form 2K for some positive integer K. The block scales the signal constellation based on how you set the Normalization method parameter
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M-ary numbe: The number of points in the signal constellation. It must have the form 2K for some positive integer K. Normalization method :Determines how the block scales the signal constellation. Choices are Min. distance between symbols, Average Power, and Peak Power.
Minimum distance :This parameter appears when Normalization method is set to Min. distance between symbols. The distance between two nearest constellation point.
Average power, referenced to 1 ohm (watts) :The average power of the symbols in the constellation, referenced to 1 ohm. This field appears only when Normalization method is set to Average Power.
Peak power, referenced to 1 ohm (watts) :The maximum power of the symbols in the constellation, referenced to 1 ohm. This field appears only when Normalization method is set to Peak Power.
Phase offset (rad):The rotation of the signal constellation, in radians. Constellation ordering:Determines how the block assigns binary words to points of the signal constellation. More details are on the reference page for the Rectangular QAM Modulator Baseband block.
The General Block Deinterleaver block rearranges the elements of its input vector without repeating or omitting any elements. If the input contains N elements, then the Elements parameter is a column vector of length N. The column vector indicates the indices, in order, of the output elements that came from the input vector. That is, for each integer k between 1 and N, Output(Elements(k)) = Input(k)
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The Elements parameter must contain unique integers between 1 and N. Both the input and the Elements parameter must be column vector signals. This block accept the following data types: int8, uint8, int16, uint16, int32, uint32, boolean, single, double, and fixed-point. The output signal inherits its data type from the input signal.
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The Matrix Deinterleaver block performs block deinterleaving by filling a matrix with the input symbols column by column and then sending the matrix contents to the output port row by row. The Number of rows and Number of columns parameters are the dimensions of the matrix that the block uses internally for its computations. This block accepts a column vector input signal. The length of the input vector must be Number of rows times Number of columns. The block accepts the following data types: int8, uint8, int16, uint16, int32, uint32, boolean, single, double, and fixed-point. The output signal inherits its data type from the input signal.
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Trellis structure :MATLAB structure that contains the trellis description of the convolutional encoder. Use the same value here and in the corresponding Convolutional Encoder block.
Punctured code :Select this check box to specify a punctured input code. The field, Punctured code, appears. Puncture vector :Constant puncture pattern vector used at the transmitter (encoder). The puncture vector is a pattern of 1s and 0s, where the 0s indicate the punctured bits. This field appears when the check box Punctured code is selected. Enable erasures input port :When you check this box, the decoder opens an input port labeled Era. Through this port, you can specify an erasure vector pattern of 1s and 0s, where the 1s indicate the erased bits. For these erasures in the incoming data stream, the decoder does not update the branch metric. The widths and the sample times of the erasure and the input data ports must be the same. The erasure input port can be of data type double or Boolean.
Decision type :Specifies the use of Unquantized, Hard Decision, or Soft Decision for the branch metric calculation.Unquantized decision uses the Euclidean distance to calculate the branch metrics. Soft Decision and Hard Decision use the Hamming distance to calculate the branch metrics, where Number of soft decision bits equals 1.
Number of soft decision bits :The number of soft decision bits used to represent each input. This field is active only when Decision type is set to Soft Decision.
Error if quantized input values are out of range :Check this box to throw an error when quantized input values are out of range. This check box is active only when Decision type is set to Soft Decision or Hard Decision.
Traceback depth :The number of trellis branches used to construct each traceback path. Operation mode :Method for transitioning between successive input frames: Continuous, Terminated, and Truncated. Enable reset input port :When you check this box, the decoder opens an input port labeled Rst. Providing a nonzero input value to this port causes the block to set its internal memory to the initial state before processing the input data.
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Receive delay Number of samples by which the received data lags behind the transmitted data. (If Tx or Rx is a vector, then each entry represents a sample.) Computation delay :Number of samples that the block should ignore at the beginning of the comparison. Computation mode :Either Entire frame, Select samples from mask, or Select samples from port, depending on whether the block should consider all or only part of the input frames.
Selected samples from frame :A vector that lists the indices of the elements of the Rx frame vector that the block should consider when making comparisons. This field appears only if Computation mode is set to Select samples from mask.
Output data :Either Workspace or Port, depending on where you want to send the output data. Variable name :Name of variable for the output data vector in the base MATLAB workspace. This field appears only if Output data is set to Workspace.
Reset port :If you check this box, then an additional input port appears, labeled Rst. Stop simulation :If you check this box, then the simulation runs only until this block detects a specified number of errors or performs a specified number of comparisons, whichever comes first.
Target number of errors :The simulation stops after detecting this number of errors. This field is active only if Stop simulation is checked. Maximum number of symbols :The simulation stops after making this number of comparisons. This field is active only if Stop simulation is checked.
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CHAPTER 9 RESULTS
Sample time = 4e-6/144 SNR value = 15 dB The result is found after simulation Error rate = 2.777e-006 Number of errors = 1 Number of bits = 3.601e + 005
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CHAPTER 10 CONCLUSION
Here we have analyzed the performance of different modulation schemes with OFDM technique in Rayleigh fading channel with gray coded bit mapping and without gray mapping. By using higher order modulation scheme (like 16- QAM, 64-QAM, 16PSK, 64-PSK) with OFDM technique in Rayleigh channel, we can transmit more data rate. In both the cases we get better performance for QAM modulation scheme. So, at this stage we can easily conclude that QAM has got better performance than PSK and use of gray bit mapping enhances this performance even more. M-ary modulation techniques provide better bandwidth efficiency than other low level modulation Techniques . As the value of M i.e. number of bits in symbol increases bandwidth utilization is increases. Also as communication range increases between a transmitter & receiver lower order modulation techniques are preferred over higher order modulation techniques . In this paper we have studied the error rate performance of different MPSK modulation schemes in normal AWGN channel & multipath Rayleigh fading channel with the help of MATLAB/Simulink, the most powerful and user friendly tool for various communication systems, digital signal processing system, control systems etc which provides easy simulation and observation of the model before it is physically made.According to the various graphs provided in this paper we can conclude that error rate is much higher in fading channel than normal AWGN channel & the error rate is further increases with the value of M i.e. number of bits in symbol increases in both AWGN & multipath fading channel. High level modulation techniques are always preferred for high data rate. As error rate increases with the value of M so low level of M-ary modulation techniques should be used for data transmission over short distance and lower level of modulation technique like QPSK should be preferred over longer distance. So to provide reliable communication along with higher data rates there should be a tradeoff between error rate & data rate.
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REFERENCES
1. John G. Proakis, Digital Communications, 4th, Mc Graw Hill, 2001. 2. Gordon L. Stuber, Principles of Mobile Communication, 2nd, Kluwer Academic Publishers, 2002. 3. Tor Aulin John B. ANderson and Carl-Erik Sundberg, Digital Phase Modulation, Kluwer Academic Publishers, 1986. 4. K.M. Chun-Hsuan Kuo, Chugg,On the bandwidth efficiency of CPM signals," Military Communications Conference, 2004. MILCOM 2004. IEEE, vol. 1, pp. 218 224, Nov. 2004. 5. Aulin T. Moqvist, P., Power and bandwidth efficient serially concatenated CPM with iterative decoding," Global Telecommunications Conference, 2000.GLOBECOM '00. IEEE, vol. 2, pp. 790 - 794, Dec. 2000. 6. Ranganathan S. Sundaravaradhan S.P. Collins O.M. Padmanabhan, K.,General CPM and its capacity," Information Theory, 2005. ISIT 2005. Proceedings. International Symposium on, pp. 750 - 754, Sept. 2005. 7. K.M. Chun-Hsuan Kuo, Chugg, The capacity of constant envelope, continuous phase signals over AWGN channel under Carson's rule bandwidth costraint," Communications, 2005. ICC 2005. 2005 IEEE International Coference on, vol. 1, pp. 218 - 224, Nov. 2004. 8. K.M. Jun Heo Kyuhyuk Chung, Chugg, Reduced state adaptive SISO algorithms for serially concatenated CPM over frequency-selective fading chanels," Global Telecommunications Conference, 2001. GLOBECOM '01. IEEE, vol. 2, pp. 11621166, Nov. 2001. 9. K.M. Chun-Hsuan Kuo, Chugg, Improving the Bandwidth Efficiency and Performance of CPM Signals via Shaping and Iterative Detection," Military Communications Conference, 2005. MILCOM 2005. IEEE, pp. 1 - 7, Oct.2005. 10. Theodore S. Pappaport, Wireless Communications, Principles and Practice, Prentice Hall, 1996. 11. C. Liang, J. Jong, W. Stark, and J. East, Non-linear amplifier effects in communications systems," Global Telecommunications Conferen2000.GLOBECOM '00. IEEE, vol. 47, no. 2, Aug. 1999
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12. A. Saleh, Frequency-Independent and Frequency-Dependent Nonlinear Mod-els of TWT Amplifiers," Communications, IEEE Transactions on, vol. 29, no.11, pp. 1715 1720, Nov. 1981. 13. J.V. Li-Chung Chang, Krogmeier, Analysis of the effects of linearity an efficiency of ampliers in QAM systems," Wireless Communications and Networking, 2003.WCNC 2003. 2003 IEEE, vol. 1, pp. 475 - 479, March 2003. 14. Tho Le-Ngoe Thai Hoa Vo, Baseband predistortion techniques for M-QAM transmission using non-linear power amplifiers," Vehicular Technology Conference, 2003. VTC 2003-Fall. 2003 IEEE 58th, vol. 1, pp. 687 - 691, Oct.2003.