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Spring 2010

**MATLAB PROJECT WORK
**

SIMPLE DIGITAL TRANSMISSION SYSTEM

1. GENERAL INFORMATION

The final deadline for returning the project work is 11.6.2010. The credit points (7 cr) will be transfered when both the exam (or mid-term exams) and the project are completed. The work should be returned to the advisor by e-mail (see the supervisor instructions below!). The e-mail should include the following items: Name, student number, and e-mail address Matlab code for the simulation(s). Your name, student number and email should be on the first line. The file name for the returned work should be your last name. (m-file attachment: your_last_name.m) Answers to the given questions (a couple of short sentences per question) In addition to the above final deadline to complete the overall task, there is also an interim review deadline on 7.5.2010 by which the Task 2.1 (described below) should be completed. Good programming discipline should be followed when writing the Matlab-code. This means that the variable names should be logical, the code must be commented and it should be written in such a way that it is easy to follow and understand. Figures should have appropriate titles and axis labels and use of axis command is recommended (see help: title, xlabel, ylabel, axis). If you have any questions concerning the work or practical arrangements, please contact the project advisor: Finnish students (TLT-5400): Ville Syrjälä TG209 E-mail: ville.syrjala@tut.fi English students (TLT-5406): Jie Zhang TG211 E-mail: jie.zhang@tut.fi

Please, use the following email-subject when returning the project: [TLT-5400/5406 Matlab Project].

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(4a ) ] T T where T is the symbol duration [s] and is the roll-off factor. “p = p/sqrt(Fs*T)”. you have to scale it properly by square-root of the over-sampling factor Fs*T (Fs denotes the sampling frequency). The block diagram of the simulation system is presented in Figure 1. make sure you use the function properly (see “help rcosine”). receive filter. Bit Source QAM Coder TX Filter Original Bits BER Computing Noise Received Bits Bit Sink QAM Decoder Detector Equalizer Sampler RX Filter Channel Figure 1. The goal in this project work is to learn and demonstrate the meaning of some essential blocks in digital transmission systems. equalizer etc. where n is an integer (e. 2/5 . TASK In this Matlab-project. Use next square root raised-cosine filter for the pulse-shaping. t t 2 p [1 . channel. After you have completed the work. Before using the pulse. you should simulate a 16-QAM transmission system (or actually a baseband equivalent transmission system since complex symbol alphabets can only be used in bandpass transmission in practice) including transmitter. Block diagram of the simulation model. Plot the eye-diagram and constellation of the generated signal. and receiver functions.e. If you do that. you can use the rcosine –function in Matlab’s Communications toolbox.g.. Note: As an alternative.The length of the signal blocks should be n*Fs*T.. 2. . See parameters from Table 1 (page 5). The roll-off factor and the over-sampling factor Fs*T are given in Table 1 at the end of this document (page 5).TLT-5400/5406 DIGITAL TRANSMISSION Spring 2010 2.a ]p ) T T T . are used and how to simulate them by using Matlab. you should know why blocks like transmit filter. . The impulse response of the filter is given as: p(t ) = 4a t t t cos([1 + a ]p ) + sin([1 . formal symbol rate to be used is Rs = 1/T = 1000 000 symbols/s (Rs = 1 MHz). i.Here it’s enough to plot the eye-diagram using only the real part of the signal (why?). n=2 or n=3).1 Transmitter Generate 10 000 random bits and map them to 16-QAM symbols.

Filter the signal with the given filter (this models the channel. Both Rx. (Re-)Sample the signal with symbol rate 1/T. you have to find out the delay of the system to sample at the correct phase (this is in practice done by a symbol timing recovery algorithm). . use. Why is that? See also Q4. . Start resampling of the filtered received signal from the position Pos. Plot the received signal in frequency domain (amplitude spectrum) after channel filtering and adding the noise.Pos] = max(abs(RxImpOut)). of course..A.NOTICE: Here (in Table 1) SNR refers to the “whole band” ratio of the signal and noise powers. % % % % Create an impulse. % % Find the maximum value and its position. In practice what matters is only the in-band noise power (i. Find out the combined impulse response of the Tx-filter.3 Receiver Filter the received signal with a filter matched to the used TX filter (square-root raised-cosine). Is it correct to use the previous channel model (time-invariant linear filter + AWGN) in modeling a radio channel in mobile transmission? Why? / Why not? Q4. Questions: Q3.e. In the simulations. your own received signal variable (called RxFiltOut here). . Questions: Q1: In general. Compare these two.Add Gaussian noise to the signal with proper SNR . “help fir1”) . you can find the delay for example in the following way: Impulse = [1 zeros(1. how can you improve the bit-error probability for a given symbol constellation (in terms of bit-to-symbol mapping)? Q2: What is the effect of the roll-off factor on the transmitted signal in general? 2.Impulse).Use logarithmic scaling for the Y-axis ( “plot(f. Before (re-)sampling. channel.ChannelImpOut). RxImpOut = filter(RxFilter.When plotting the spectra.e.Use correct scaling for the frequency axis. % % RxSampOut = RxFiltOut(Pos:Fs*T:end)). Fcut is the cut-off frequency for the channel model filter (values for the SNR and Fcut from Table 1). include both negative and positive frequencies (i. TxImpOut = filter(TxFilter. Plot the eye-diagram of the received signal after RX filter. and Rx-filter.100)].2 Transmission over Channel Simulate the channel model (linear filter and additive white Gaussian noise.. –Fs/2 … Fs/2). AWGN. the power of the noise within the signal band).TxImpOut). 3/5 . [Trash.and Tx-filters are square-root raised-cosine filters.TLT-5400/5406 DIGITAL TRANSMISSION Spring 2010 Examine the spectral content of the transmitted signal by plotting the amplitude spectra of (i) the transmit filter and (ii) the actual transmitted signal. In the actual resampling. ChannelImpOut = filter(B. What is roughly the in-band SNR in the signal entering the receiver? 2. 20*log10(abs(H)))” ). . see the block-diagram in Figure 1) with given SNR and bandwidth.1. see “help butter”.1. now the position is the delay.

TLT-5400/5406 DIGITAL TRANSMISSION Spring 2010 Plot the symbol rate constellation of the sampled signal (i. the whole received sample sequence should be filtered with the inverse response (with the equalizer impulse response). 4/5 . include the noise and tune the algorithm if necessary. The number of iterations and the order/length of the equalizer depend on the situation. try changing the iteration step-size (usually increase it)..Here it is enough to design the equalizer using only the real part of the received samples (why?). What is the meaning of the eye-diagram? Q6. After having a properly working equalizer. plot the constellation of the equalized signal.e. Intuitively. 100 000) to get more reliable BER estimates.e. and plot the obtained BER values as a function of the varied parameter.. if the last digit of your student number is 0. Questions: Q5. Then decode the detected symbols to bits and compute the bit error rate (BER). 300 kHz.g.. matched filter to the transmitted pulse? Q7. the total number of simulation runs is 4.For example. Explain what you see. good values to start with are. the complex samples after receiver filtering and resampling). Use LMS algorithm-based equalizer for the channel equalization. try to get the equalizer to work without noise. First.g. Detect the received equalized samples using a symbol-by-symbol minimum distance detector. . Then. If you are interested :o). Idea of matched filtering in general? True matched filter (matched to the received pulse) vs. i. depending on the last digit of your student number (see Table 1). What is the idea of equalizer? Some comments: In modeling the channel linear distortion. .. If you have problems with the convergence of the LMS algorithm. and 500 kHz. When the inverse response is found. e. you try to find the inverse response (roughly) of the effective channel by using a known training sequence. number of iterations. the used functions (fir1 and butter) are really simple and yield channel models (with the given parameters) that are “reasonably” easy to equalize. 1000-5000 iterations and the order/length between 10-30. you should run your simulation with the parameter SNR = 15 dB (fixed) and Fcut = 200 kHz. 400 kHz. and/or the equalizer order/length. Repeat the simulation for different channel cut-off frequencies Fcut or different SNR’s. repeat the simulations with higher number of bits (e. Remember to check correct parameters from Table 1.

25 Range 200 – 500 5 – 20 5 – 20 100 – 400 5 – 20 5 – 20 200 – 500 5 – 20 200 – 500 200 – 500 Step 100 5 5 100 5 5 100 5 100 100 5/5 .30 0. Symbol rate 1/T = 1 MHz is a common parameter for everyone.25 0.25 0.35 0. Fs 4 MHz 3 MHz 2 MHz 4 MHz 3 MHz 2 MHz 4 MHz 3 MHz 2 MHz 2 MHz Over-sampling Factor FsT 4 3 2 4 3 2 4 3 2 2 Rolloff Channel Model (filter) 3rd order Butterworth 4th order FIR 4th order FIR 12th order FIR 3rd order Butterworth 5th order FIR 2nd order Butterworth 3rd order Butterworth 3rd order Butterworth 3rd order Butterworth Plot BER as a Function of Fcut [kHz] SNR [dB] SNR [dB] Fcut [kHz] SNR [dB] SNR [dB] Fcut [kHz] SNR [dB] Fcut [kHz] Fcut [kHz] Fixed Parameter SNR = 15 dB Fcut = 500 kHz Fcut = 500 kHz SNR = 10 dB Fcut = 500 kHz Fcut = 500 kHz SNR = 15 dB Fcut = 500 kHz SNR = 15 dB SNR = 15 dB α 0.35 0. Simulation parameters which depend on the last digit of your student number.TLT-5400/5406 DIGITAL TRANSMISSION Spring 2010 Table 1.40 0. Last Digit of Student # 0 1 2 3 4 5 6 7 8 9 Sampling Freq.30 0.35 0.30 0.

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