ECE 146A/UCSB: Introduction to Communication Systems

Lab 5: Introduction to Digital Modulation

This is the first lab in a sequence that develops a reasonably complete matlab simulator for a
linearly modulated digital communication system.

Background

Figure 1 shows block diagrams corresponding to a typical DSP-centric realization of a communi-

cation transceiver employing linear modulation. In the labs, we model the core components of such
a system using the complex baseband representation, as shown in Figure 2. Given the equivalence
of passband and complex baseband, we are only skipping modeling of finite precision effects due
to digital-to-analog conversion (DAC) and analog-to-digital conversion (ADC). These effects can
easily be incorporated into Matlab models such as those we will develop, but are beyond the scope
of the current set of labs.
I I
Two−dimensional Transmit DAC
symbols filter
(implemented Upconverter
rate 1/T Q Q
at rate 4/T) DAC

Digital Analog Passband

streams baseband Channel
rate 4/T waveforms

I I
DSP for Receive ADC
Estimated filter
receiver (implemented Dnconverter
symbols Q Q
functions at rate 4/T) ADC
(includes coarse analog
(synchronization, Digital Analog passband filtering)
equalization, streams baseband
demodulation) rate 4/T waveforms

Figure 1: Typical DSP-centric transceiver realization. Our model does not include the blocks
shown in dashed lines. Finite precision effects such as DAC and ADC are not considered. The
upconversion and downconversion operations are not modeled. The passband channel is modeled
as an LTI system in complex baseband.

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 Sampler,
rate m/T
Symbols Transmit Receive
Channel Receiver Estimated
{b[n]} Filter Filter Filter Signal
Rate 1/T g (t) g (t) g RX(t) Processing symbols
TX C

(Synchronization,
Noise Equalization,
Demodulation)

Figure 2: Block diagram of a linearly modulated system, modeled in complex baseband.

A few points worth noting about the model of Figure 2:

Choice of transmit filter: The PSD of the transmitted signal is proportional to |GT X (f )|2
(see Chapter 4). The choice of transmit filter is made based on spectral constraints, as well as
considerations such as sensitivity to receiver timing errors and intersymbol interference. Typically,
the bandwidth employed is of the order of T1 .
Channel model: We typically model the channel as an linear time-invariant (LTI) system. For
certain applications, such as wireless communications, the channel may be modeled as slowly time
varying.
Noise model: Noise is introduced in a later lab.
Receive filter and sampler: The optimal choice of receive filter is actually a filter matched to
the cascade of the transmit filter and the channel. In this case, there is no information loss in
sampling the output of the receive filter at the symbol rate T1 . Often, however, we use a suboptimal
choice of receive filter (e.g., a wideband filter flat over the signal band, or a filter matched to the
transmit filter). In this case, it is typically advantageous to sample faster than the symbol rate.
In general, we assume that the sampler operates at rate m T , where m is a positive integer. The
output of the sampler is then processed, typically using digital signal processing (DSP), to perform
receiver functions such as synchronization, equalization and demodulation..
The simulation of a linearly modulated system typically involves the following steps.
Step 1: Generating random symbols to be sent
We restrict attention in this lab to Binary Phase Shift Keying (BPSK). Here, the symbols {bn }
in Figure 1 take values ±1.
Step 2: Implementing the transmit, channel, and receive filters
Since the bandwidth of these filters is of the order of T1 , they can be accurately implemented
in DSP by using FIR filters operating on samples at a rate which is a suitable multiple of T1 . The
default choice of sampling rate in the labs is T4 , unless specified otherwise. If the filter is specified
in continuous time, typically, one simply samples the impulse response at rate T4 , taking a large
enough filter length to capture most of the energy in the impulse response.
Step 3: Sending the symbols through the filters.
To send symbols at rate T1 through filters implemented at rate T4 , it is necessary to zeropad
(or up-sample) the symbols before convolving them with the filter impulse response determined in
Step 2.
Step 4: Adding noise
Typically, we add white Gaussian noise (model to be specified in a later lab) at the input to

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the receive filter.
Step 5: Processing at the receive filter output
If there is no intersymbol interference (ISI), the processing simply consists of sampling at rate
1
T to get decision statistics for the symbols of interest. For BPSK, you might simply take the sign
of the decision statistic to make your bit decision.
If the ISI is significant, then channel equalization (discussed in later labs) is required prior to
making symbol decisions.

Laboratory Assignment

0) Get the matlab code needed (lab5init) to get started from the course home page, and make
sure you understand what it is doing. For your information, the transmit filter specified in the code
is approximately a square root raised cosine pulse in the frequency domain.
1) Plot the impulse response of the transmit filter versus t/T .
2) Using the DFT, compute the magnitude of the transfer function of the transmit filter versus
the normalized frequency f T . (See Example 2.5.4 in Chapter 2 for a discussion of how to use
the DFT of the sampled response to estimate the Fourier transform of the original continuous
time signal that the samples correspond to). From eyeballing the plot, estimate the normalized
bandwidth (i.e., bandwidth as a multiple of T1 ).
3) Plot the response at the output of the receive filter to a single symbol. Is the transmit filter
(approximately) square root Nyquist at rate 1/T ?
4) Generate 100 random bits {a[n]} taking values in {0, 1}, and map them to symbols {b[n]}
taking values in {−1, +1}, with 0 mapped to +1 and 1 to −1.
5) Send the 100 symbols {b[n]} through the system. What is the length of the corresponding
output of the transmit filter? What is the length of the corresponding output of the receive filter?
Plot separately the input to the receive filter, and the output of the receive filter versus time, with
one unit of time on the x-axis equal to the symbol time T .
6) Do the best job you can in recovering the transmitted bits {a[n]} by directly sampling the
input to the receive filter, and add lines in the matlab code for implementing your idea. That
is, select a set of 100 samples, and estimate the 100 transmitted bits based on the sign of these
samples. (What sampling delay and spacing would you use?). Estimate the probability of error
(note: no noise has been added).
7) Do the best job you can in recovering the transmitted bits by directly sampling the output
of the receive filter, and add lines in the matlab code for implementing your idea. That is, select a
set of 100 samples, estimate the 100 transmitted bits based on the sign of these samples. (What
sampling delay and spacing would you use?). Estimate the probability of error. Also estimate the
probability of error if you chose an incorrect delay, offset from the correct delay by T /2.

Attempt parts 8) and 9) only if you have time.

8) Suppose that the receiver LO used for downconversion is ahead in frequency and phase relative
1
to the incoming wave by ∆f = 25T and a phase of π/2. Modify your complex baseband model to
include the effects of the carrier phase and frequency offset. When you now sample at the “correct”
delay as determined in 7), do a scatter plot of the complex-valued samples {y[n], n = 1, ..., 100}
that you obtain. Can you make correct decisions based on taking the sign of the real part of the
samples, as in 7)?

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 9) Now consider a differentially encoded system in which we send {a[n], n = 1, ..., 99}, where
a[n] ∈ {0, 1}, by sending the following ±1 bits: b[1] = +1, and for n = 2, ..., 100

b[n − 1], a[n] = 0,
b[n] =
−b[n − 1], a[n] = 1,

Devise estimates for the bits {a[n]} from the samples {y[n]} in 8), and estimate the probability of
error.
Hint: What does y[n]y ∗ [n − 1] look like?
Lab Report: Your lab report should answer the preceding questions in order, and should document
the reasoning you used and the difficulties you encountered. Comment on whether you get better
error probability in 6) or 7), and why?