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Lab 7: Amplitude Modulation with Suppressed Carrier

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ECEN 4652/5002 Communications <strong>Lab</strong> Spring 2013<br />

3-06-13 P. Mathys<br />

<strong>Lab</strong> 7: <strong>Amplitude</strong> <strong>Modulation</strong> <strong>with</strong> <strong>Suppressed</strong> <strong>Carrier</strong><br />

1 Introduction<br />

Many channels either cannot be used to transmit baseband signals at all, or pass signal<br />

energy very inefficiently, except for a relatively narrow passband region at frequencies substantially<br />

higher than those contained in a baseband message signal. A well-known example<br />

is electromagnetic transmission of radio signals at a frequency fc in free space which requires<br />

an antenna of length comparable to λc/2 for a dipole, or λc/4 for a monopole, where<br />

λc = 3×10 8 /fc is the wavelength in meters corresponding to fc in Hz. Thus, transmission at<br />

fc = 10 kHz would require an antenna of length comparable to 15 km for a dipole, whereas<br />

at fc = 900 MHz a length of 8.3 cm is enough for the monopole antenna of a cell phone.<br />

1.1 <strong>Amplitude</strong> <strong>Modulation</strong> <strong>with</strong> <strong>Suppressed</strong> <strong>Carrier</strong><br />

The most straightforward way to shift a signal spectrum from baseband to a passband<br />

location <strong>with</strong> center frequency fc is to make use of the frequency shift property of the<br />

Fourier transform (FT) which says that<br />

m(t) e j(2πfct+θc)<br />

⇐⇒ M(f − fc) e jθc .<br />

Thus, Ac m(t) e j(2πfct+θc) is a complex-valued bandpass signal <strong>with</strong> amplitude Ac and center<br />

frequency fc if m(t) is a (bandlimited) baseband signal. To make this into a real bandpass<br />

signal x(t), write<br />

x(t) = Re{Ac m(t) e j(2πfct+θc) } = Re Ac m(t) cos(2πfct + θc) + j sin(2πfct + θc) <br />

= Ac m(t) cos(2πfct + θc) ,<br />

where for the last equality it is assumed that m(t) is real-valued. The signal x(t) obtained in<br />

this way is a AM-DSB-SC (amplitude modulation, double side-band, suppressed carrier)<br />

signal <strong>with</strong> carrier frequency fc, carrier phase θc and Fourier transform<br />

x(t) = Ac m(t) cos(2πfct + θc) ⇐⇒ X(f) = Ac <br />

M(f − fc) e<br />

2<br />

jθc + M(f + fc) e −jθc .<br />

In the frequency domain this can be visualized as follows (setting θc = 0)<br />

Mm<br />

M(f)<br />

−fm 0 fm<br />

f<br />

AcMm<br />

2<br />

−fc<br />

−fc−fm −fc+fm<br />

1<br />

0<br />

X(f)<br />

fc−fm<br />

LSB<br />

USB<br />

<br />

fc<br />

fc+fm<br />

f


From the figure it is evident that if the bandwidth of m(t) is fm, then the bandwidth of x(t)<br />

is 2fm, which explains the “DSB” in AM-DSB-SC. It is also clear that if m(t) has no dc<br />

component (which is the case for speech and music signals, for instance), then x(t) has no<br />

component at the carrier frequency fc, which is where the “SC” comes from. The portion<br />

of the spectrum of x(t) for which fc − fm ≤ |f| < fc is called the lower side-band (LSB),<br />

whereas the portion for which fc < |f| ≤ fc + fm is called the upper side-band (USB).<br />

To recover m(t) undistorted from x(t), fc ≥ fm is required, but usually fc ≫ fm in practice.<br />

The block diagram of an AM-DSB-SC transmission system is shown in the following figure.<br />

Transmitter<br />

<br />

mw(t) LPF m(t) x(t) Channel<br />

r(t) v(t) LPF ˆm(t)<br />

× + ×<br />

at fm<br />

HC(f)<br />

at fL<br />

Ac cos(2πfct + θc)<br />

<strong>Carrier</strong> Oscillator<br />

Noise n(t)<br />

<br />

Channel Model<br />

Receiver<br />

<br />

2cos(2πfct + θc)<br />

Local Oscillator<br />

The transmitter consists of a LPF that bandlimits the wideband message signal mw(t) to<br />

|f| ≤ fm and the modulator which multiplies the resulting message signal m(t) <strong>with</strong> the<br />

output Ac cos(2πfct + θc) of the carrier oscillator. The channel is modeled as a filter HC(f)<br />

<strong>with</strong> noise added at the output. In the receiver the incoming signal r(t) is multiplied by<br />

the local oscillator signal cos(2πfct + θc) and then lowpass filtered at fL. Assuming an ideal<br />

channel <strong>with</strong> attenuation γ and no noise such that r(t) = γ x(t), the demodulation operation<br />

can be described as<br />

v(t) = 2r(t) cos(2πfct+θc) = 2γAcm(t) cos 2 (2πfct+θc) = γAcm(t) 1 + cos(4πfct+2θc) .<br />

Assuming that fc ≥ fm, the second term, which is a AM-DSB-SC signal <strong>with</strong> carrier frequency<br />

2fc and carrier phase 2θc, can be removed by lowpass filtering at fL = fm and<br />

thus<br />

ˆm(t) = γAc m(t) .<br />

In the absence of noise and other channel impairments this is an exact replica of the transmitted<br />

message signal, scaled by γAc.<br />

If m(t) is a wide-sense stationary process <strong>with</strong> mean E[m] and autocorrelation function<br />

Rm(τ), then the autocorrelation function of the AM-DSB-SC signal x(t) can be computed<br />

as<br />

Rx(t1, t2) = E Ac m(t1) cos(2πfct1 + θc) A ∗ c m ∗ (t2) cos(2πfct2 + θc) <br />

= |Ac| 2 E[m(t1) m ∗ (t2)]<br />

<br />

= Rm(t1 − t2)<br />

= |Ac| 2<br />

cos(2πfct1 + θc) cos(2πfct2 + θc)<br />

<br />

<br />

<br />

cos 2πfc(t1 − t2) + cos <br />

2πfc(t1 + t2) + 2θc<br />

= 1<br />

2<br />

2 Rm(t1 − t2) cos 2πfc(t1 − t2) + cos <br />

2πfc(t1 + t2) + 2θc .<br />

2


Note that x(t) is a cyclostationary process <strong>with</strong> period 1/fc. The time-averaged autocorrelation<br />

function of x(t) is<br />

¯Rx(τ) = fc<br />

1/fc<br />

0<br />

Rx(t + τ, t) dt =<br />

|Ac| 2<br />

2 Rm(τ) cos(2πfcτ) .<br />

Thus, if m(t) has PSD Sm(f), then the PSD of the AM-DSB-SC signal x(t) is<br />

Sx(f) =<br />

2 |Ac| <br />

Sm(f − fc) + Sm(f + fc)<br />

4<br />

.<br />

The PSD of a speech signal after AM-DSB-SC modulation <strong>with</strong> fc = 8000 Hz and fm = 4000<br />

Hz is shown in the following graph.<br />

10log 10 (S x (f)) [dB]<br />

0<br />

−10<br />

−20<br />

−30<br />

−40<br />

−50<br />

PSD, P x =0.0072139, P x (f 1 ,f 2 ) = 49.9999%, F s =44100 Hz, N=44100, NN=3, Δ f =1 Hz<br />

−60<br />

0 2000 4000 6000 8000<br />

f [Hz]<br />

10000 12000 14000 16000<br />

1.2 Coherent AM Reception<br />

An idealizing assumption which is tacitly made in the AM-DSB-SC transmission system<br />

block diagram given earlier, is that the local oscillator at the receiver is synchronized <strong>with</strong><br />

the carrier oscillator at the transmitter. To see why this synchronism between transmitter<br />

and receiver is important, assume that the local oscillator signal is 2 cos(2πfct), but the<br />

received AM-DSB-SC signal is r(t) = γAcm(t) cos(2πfct + θe), i.e., there is a phase error θe<br />

between transmitter and receiver. Now the receiver computes<br />

and thus<br />

v(t) = 2γAc m(t) cos(2πfct + θe) cos(2πfct) = γAc m(t) cos θe + cos(4πfct + θe) ,<br />

ˆm(t) = γAc cos θe m(t) ,<br />

after the LPF at fL = fm. A small phase error |θe| ≪ π/2 therefore attenuates m(t) by<br />

cos(θe) ≈ 1, which presents no big problem, but phase errors close to ±90 ◦ attenuate m(t)<br />

substantially or even suppress it altogether. This is especially annoying when θe varies <strong>with</strong><br />

time and ˆm(t) changes periodically in intensity. On the positive side, however, this means<br />

that two AM-DSB-SC signals, such as<br />

xi(t) = Acmi(t) cos(2πfct) , and xq(t) = Acmq(t) cos(2πfct + π/2) ,<br />

3


can use the same carrier frequency fc to transmit two independent message signals mi(t)<br />

and mq(t). This is known as quadrature amplitude modulation (QAM), and xi(t)<br />

is called the in-phase component of the AM signal at fc, whereas xq(t) is called the<br />

quadrature component. At any rate, it is crucial for the correct demodulation of AM<br />

signals <strong>with</strong> suppressed carrier, that the receiver is phase (and frequency) synchronized<br />

<strong>with</strong> the transmitter. Receivers of this type are called synchronous or coherent receivers.<br />

In practice the maintenance of exact phase synchronism between two oscillators in different<br />

physical locations is quite a non-trivial problem and requires a considerable amount of active<br />

hardware and/or software.<br />

1.3 AM-SSB-SC and AM-VSB-SC<br />

One of the disadvantages of AM-DSB-SC is that it occupies twice the bandwidth of the<br />

original message signal. One straightforward way to reduce the bandwidth to the original<br />

value is to only keep one of the sidebands of the AM signal and suppress the other one. The<br />

resulting AM signals are known as AM-SSB-LSB (amplitude modulation, single sideband,<br />

lower sideband) and as AM-SSB-USB (amplitude modulation, single sideband, upper sideband)<br />

depending on whether the lower or upper sideband is kept. To convert AM-DSB-SC<br />

to AM-SSB-SC (either LSB or USB), the AM-DSB-SC signal can be filtered <strong>with</strong> a bandpass<br />

filter (BPF) as shown in the following block diagram.<br />

mw(t) LPF m(t) x(t) BPF xB(t)<br />

×<br />

at fm<br />

HBx(f)<br />

Ac cos(2πfct + θc)<br />

<strong>Carrier</strong> Oscillator<br />

For AM-SSB-USB, for example, the transmitter filter HBx(f) is chosen as shown in the<br />

following figure.<br />

Filter for AM-SSB-USB<br />

1<br />

HBx(f)<br />

−fc−fm −fc −fc+fm 0 fc−fm fc fc+fm<br />

A problem <strong>with</strong> this filter are the sharp cutoffs needed near fc, especially if m(t) has a dc<br />

component (which is the case for analog TV broadcast signals, for instance). To alleviate<br />

this problem, vestigial sideband (VSB) modulation can be used. This is essentially a<br />

compromise between AM-DSB and AM-SSB, <strong>with</strong> a well controlled (usually linear) overall<br />

transition from the passband of HB(f) to the stopband near fc, extending over a range of<br />

2∆ around fc. Depending on whether the lower or upper sideband is kept, the resulting<br />

4<br />

f


AM signal is either called AM-VSB-LSB (amplitude modulation, vestigial sideband, lower<br />

sideband) or AM-VSB-USB (amplitude modulation, vestigial sideband, upper sideband).<br />

An example of a filter HB(f) that converts a AM-DSB-SC signal to a AM-VSB-USB-SC<br />

signal is shown in the following figure.<br />

Filter for AM-VSB-USB<br />

−fc−fm −fc −fc+fm<br />

−fc−∆ −fc+∆<br />

1<br />

HB(f)<br />

0 fc−fm fc fc+fm<br />

fc−∆ fc+∆<br />

Demodulation of AM-SSB-SC signals and AM-VSB-SC signals is done in a similar fashion<br />

as for AM-DSB-SC by multiplying the received signal <strong>with</strong> the local oscillator signal<br />

2 cos(2πfct + θc), followed by lowpass filtering at fm. To remove noise and/or interference<br />

from the unused (portion of the) sideband, a BPF should be used at the input of the receiver,<br />

as shown in the following blockdiagram.<br />

r(t) BPF rB(t) v(t) LPF ˆm(t)<br />

×<br />

HBr(f)<br />

at fL<br />

2cos(2πfct + θc)<br />

Local Oscillator<br />

For AM-SSB-SC the same BPF can be used for both the transmitter and the receiver. For<br />

AM-VSB-SC the product HBx(f)HBr(f) of the frequency responses of the BPFs at the<br />

transmitter and receiver must be equal to HB(f) as shown above.<br />

1.4 Bandpass Filters<br />

Suppose you have a lowpass filter hL(t) ⇔ HL(f), e.g., an LPF <strong>with</strong> trapezoidal frequency<br />

response and thus<br />

hL(t) = sin(2πfLt)<br />

πt<br />

sin(2παfLt)<br />

2παfLt<br />

⇐⇒<br />

−(1+α)fL<br />

−(1−α)fL<br />

5<br />

1<br />

0<br />

HL(f)<br />

(1−α)fL<br />

(1+α)fL<br />

0 ≤ α ≤ 1<br />

f<br />

f


By making use of the frequency shift property of the FT, this LPF can be converted to a<br />

BPF hBP (t) ⇔ HBP (f) which is symmetric about some center frequency fc ≥ (1 + α) fL<br />

(where α = 0 for an ideal LPF) by<br />

hBP (t) = 2 hL(t) cos(2πfct) ⇐⇒ HBP (f) = HL(f) ∗ [δ(f − fc) + δ(f + fc)] .<br />

BPFs that are obtained from ideal LPFs (i.e., α → 0) are well suited for picking out one<br />

particular signal from several FDM (frequency division multiplexed) signals, or for generating<br />

SSB (single sideband) AM signals from DSB AM signals. BPFs that are obtained from LPFs<br />

<strong>with</strong> trapezoidal frequency response can be used for similar tasks, but in addition they can<br />

also be used to convert frequency to amplitude (in the transition region of the BPF) and to<br />

generate VSB (vestigial sideband) AM signals.<br />

1.5 <strong>Carrier</strong> Frequency Extraction<br />

Let r(t) be a received noiseless AM-DSB-SC signal <strong>with</strong> attenuation γ, i.e.,<br />

r(t) = γ x(t) = γAc m(t) cos <br />

2π(fc + fe)t + θe ,<br />

where fe is the frequency error and θe is the phase error between the transmitter and the<br />

receiver. To obtain (an estimate of) the error signal ψ(t) = 2πfet + θe from r(t), start from<br />

squaring r(t) to obtain<br />

r 2 (t) = γ 2 A 2 c m 2 (t) cos 2 γ<br />

2π(fc + fe)t + θe = 2 A2 c m2 (t) <br />

1 + cos 4π(fc + fe)t + 2θe .<br />

2<br />

Multiplying this by 2 cos(4πfct) yields<br />

vi(t) = γ 2 A 2 c m 2 (t) 1 + cos <br />

4π(fc + fe)t + 2θe cos 4πfct<br />

= A(t) 2 cos 4πfct + cos(4πfet + 2θe) + cos <br />

4π(2fc + fe)t + 2θe ,<br />

where A(t) = γ 2 A 2 c m 2 (t)/2 is a time-varying amplitude. Simlarly, multiplying by -2 sin 4πfct<br />

results in<br />

vq(t) = −γ 2 A 2 c m 2 (t) 1 + cos <br />

4π(fc + fe)t + 2θe sin 4πfct<br />

= A(t) − 2 sin 4πfct + sin(4πfet + 2θe) − sin <br />

4π(2fc + fe)t + 2θe .<br />

Thus, after lowpass filtering <strong>with</strong> 2fe < fL < fc,<br />

wi(t) = A(t) cos(4πfet + 2θe) and wq(t) = A(t) sin(4πfet + 2θe) .<br />

Finally, the error estimate ψ(t) is obtained by taking an inverse tangent and dividing by 2<br />

as follows<br />

ψ(t) = 1<br />

2 tan−1<br />

<br />

wq(t)<br />

<br />

.<br />

wi(t)<br />

This whole process is shown in blockdiagram form in the next figure.<br />

6


(t)<br />

(.) 2<br />

r 2 (t)<br />

•<br />

×<br />

2cos4πfct<br />

×<br />

vi(t)<br />

vq(t)<br />

−2sin4πfct<br />

LPF<br />

at fL<br />

LPF<br />

at fL<br />

wi(t)<br />

tan −1 wq(t)<br />

wi(t)<br />

wq(t)<br />

Note that, before the division by 2 to obtain ψ(t), it is crucial that the phase (which is<br />

only resolved modulo 2π by the inverse tangent) is unwrapped. To demodulate the received<br />

AM-DSB-SC signal r(t), the local oscillator term 2 cos(2πfct + ψ(t)) is then used instead of<br />

the 2 cos(2πfct + θc) term shown in an earlier blockdiagram.<br />

2 <strong>Lab</strong> Experiments<br />

E1. AM-DSB-SC Transmitter/Receiver. (a) Write a Matlab function, called amxmtr10<br />

which performs the tasks of a AM-DSB-SC transmitter, i.e., bandlimits the message signal to<br />

fm (using trapfilt) and then multiplies the result <strong>with</strong> the output of the carrier oscillator.<br />

The header of amxmtr10 looks as follows:<br />

function x = amxmtr10(t,m,fcparms,fmparms)<br />

%amxmtr10 <strong>Amplitude</strong> <strong>Modulation</strong> Transmitter V1.0<br />

% >>>>> x = amxmtr10(t,m,fcparms,fmparms)


Test your transmitter using the message signal<br />

Fs = 44100; %Sampling rate<br />

t = [0:Fs-1]/Fs; %Time axis<br />

m = (cos(2*pi*3000*t)+cos(2*pi*5000*t)); %Message signal<br />

as input. Set fc = 8000 Hz, θc = 0 ◦ , fm = 4000, km ≈ 10 . . . 20, and αm = 0.05. The LPF<br />

at the transmitter should remove the frequency component at 5000 Hz. The 3000 Hz cosine<br />

should be moved to fc ± 3000 Hz so that the PSD looks as shown below.<br />

S x (f)<br />

0.07<br />

0.06<br />

0.05<br />

0.04<br />

0.03<br />

0.02<br />

0.01<br />

PSD, P x =0.2498, P x (f 1 ,f 2 ) = 0.1249, F s =44100 Hz, N=44100, NN=1, Δ f =1 Hz<br />

0<br />

0 2000 4000 6000 8000<br />

f [Hz]<br />

10000 12000 14000 16000<br />

(b) Use the speech signal in speech701.wav and the music signal in music701.wav to<br />

generate AM-DSB-SC signals x1(t) and x2(t), respectively, <strong>with</strong> fc = 8000 Hz, fm = 4000<br />

Hz, km ≈ 10 . . . 20, and αm = 0.05. Use θc = −90 ◦ for the speech signal and θc = 0 ◦ for<br />

the music signal. Create a third signal x3(t) = (x1(t) + x2(t))/2. Look at the PSDs of<br />

each of the three signals. Does the bandwidth for x3(t), which contains two message signals,<br />

change? Also look at the PSDs of the squared AM signals x 2 1(t), x 2 2(t), and x 2 3(t). Is there<br />

any additional information that you can get from the squared signals? If so, what is this<br />

information and for which of the three signals is it actually present? Save the three signals<br />

for later use.<br />

(c) Write a Matlab function called amrcvr10 that demodulates a received AM-DSB-SC<br />

signal r(t) and produces an estimate ˆm(t) of the transmitted message m(t). Here is the<br />

header for this function<br />

8


function mhat = amrcvr10(t,r,fcparms,fmparms)<br />

%amrcvr10 <strong>Amplitude</strong> <strong>Modulation</strong> Receiver, V1.0<br />

% >>>>> mhat = amrcvr10(t,r,fcparms,fmparms)


To test your modified trapfilt function, estimate the parameters of the BPF whose frequency<br />

response is shown below and recreate h(t) ⇔ H(f) <strong>with</strong> your trapfilt function.<br />

∠X(f) [deg]<br />

|X(f)|<br />

1.4<br />

1.2<br />

1<br />

0.8<br />

0.6<br />

0.4<br />

0.2<br />

200<br />

100<br />

−100<br />

FT Approximation , F s =44100 Hz, N=44100, Δ f =1 Hz<br />

0<br />

0 2000 4000 6000 8000 10000 12000 14000 16000<br />

0<br />

−200<br />

0 2000 4000 6000 8000<br />

f [Hz]<br />

10000 12000 14000 16000<br />

(b) Extend the Matlab functions amxmtr10 and amrcvr10 so that they can also be used<br />

transmit and receive AM-SSB-SC and AM-VSB-SC. Call the new functions amxmtr11 and<br />

amrcvr11. The header of amxmtr11 is:<br />

10


function x = amxmtr11(t,m,fcparms,fmparms,fBparms)<br />

%amxmtr11 <strong>Amplitude</strong> <strong>Modulation</strong> Transmitter, V1.1<br />

% >>>>> x = amxmtr11(t,m,fcparms,fmparms,fBparms)


10log 10 (S x (f)) [dB]<br />

0<br />

−10<br />

−20<br />

−30<br />

−40<br />

−50<br />

PSD, P x =0.0026165, P x (f 1 ,f 2 ) = 49.9999%, F s =44100 Hz, N=44100, NN=3, Δ f =1 Hz<br />

−60<br />

0 2000 4000 6000 8000<br />

f [Hz]<br />

10000 12000 14000 16000<br />

The header of the receiver function amrcvr11 is:<br />

function mhat = amrcvr11(t,r,fcparms,fmparms,fBparms)<br />

%amrcvr11 <strong>Amplitude</strong> <strong>Modulation</strong> Receiver, V1.1<br />

% >>>>> mhat = amrcvr11(t,r,fcparms,fmparms,fBparms)


signals, and demodulating them <strong>with</strong> the AM receiver function. Use Fs = 44100 Hz, fm =<br />

4000 Hz, fc = 8000 Hz, θc = 0 ◦ , and, in the case of AM-VSB, ∆ = 1000 Hz. Look at the<br />

PSDs Sx(f) of all four AM signals x(t) and also at the PSDs of x 2 (t). Are they what you<br />

expect them to be? If not, why not?<br />

(d) Analyze and demodulate the signals in the wav-files amsig704.wav and amsig705.wav.<br />

Determine the important parameters such as fc, θc, SSB or VSB, USB or LSB, fm and,<br />

in the case of VSB, ∆. Which parameters can you obtain from PSDs? For which ones do<br />

you actually have to demodulate the received signals? Note that the signals may have noise<br />

and/or interference from other signals added. If this is the case, try to eliminate as much of<br />

the noise/interference as possible. Explain your strategy!<br />

(e) Revisit those AM signals in the files amsig701.wav, amsig702.wav, and amsig703.wav<br />

that you could not demodulate properly in E1(d). Try if using an AM-SSB receiver (which<br />

removes one of the sidebands before demodulation) helps. If so, can you explain why?<br />

E3. Extraction of <strong>Carrier</strong> Frequency and Phase from AM-DSB-SC Signal. (Experiment<br />

for ECEN 5002, optional for ECEN 4652) (a) Modify your amrcvr11 receiver<br />

function so that it extracts the error estimate ψ(t) = 2πfet + θe between the carrier oscillator<br />

at the transmitter (cos <br />

2π(fc + fe)t + θe ) and the local oscillator at the receiver<br />

(cos 2πfct). Use this to demodulate AM-DSB-SC signals <strong>with</strong> unknown carrier phase and/or<br />

unstable carrier. Here is the header of the new function, called amrcvr12.<br />

13


function mhat = amrcvr12(t,r,fcparms,fmparms,fBparms,fxparms)<br />

%amrcvr11 <strong>Amplitude</strong> <strong>Modulation</strong> Receiver, V1.2 <strong>with</strong> automatic<br />

% carrier extraction for AM-DSB-SC<br />

% >>>>> mhat = amrcvr12(t,r,fcparms,fmparms,fBparms,fxparms)

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