Some self-study material (ELI revision)
Signal transmission through a linear system
Ideal vs practical filters
Signal distortion over a channel
Energy spectral density
Power spectral density
The Discrete Fourier Transform (DFT)
Chapter 4
Sections 3.1-3.3:
3.1 Aperiodic signal representation by Fourier integral
3.2 Transforms of some useful functions
3.3 Properties of the Fourier transform
LTI Systems:
Output in the time domain: y(t) = h(t) * x(t)
Output in the frequency domain: Y(f) = H(f)X(f)
A channel can be modeled as an LTI system.
X(f) - Fourier transform of channel input x(t)
Y(f) - Fourier transform of channel output y(t)
H(f) - Fourier transform of channel impulse response h(t)
Output of LTI (polar form): |Y(f)|e^{jθy(f)} = |H(f)||X(f)|e^{j[θx(f) + θ_h(f)]}
Distortionless channel:
|H(f)| = k
θh(f) = -2πftd
Where t_d is the channel time delay (same for all f).
Difference between an all-pass and distortionless channel.
Ideal causal filter:
y(t) = g(t - t_d)
h(t) = F^{-1}{H(f)} = 2Bsinc[2B(t - t_d)]
Is impulse response realisable?
How can we make a realisable filter?
A necessary and sufficient condition for |H(f)| to be a causal filter:
\int_{-\infty}^{\infty} \frac{|\ln |H(f)||}{1 + (2\pi f)^2} df < \infty
|H(f)| = 0 in a band?
|H(f)| = 0 at a single frequency?
Linear Channel Distortion:
Amplitude response distortion: |H(f)| ≠ k
Phase response distortion: |θh(f)| ≠ -2πftd
Results in a non-ideal channel.
Results in dispersion (analog) or intersymbol interference (digital).
Pulse is dispersed when it passes through a system that is not distortionless.
A low-pass filter transfer function H(f) is given by:
H(f) = \begin{cases} (1 + k \cos 2\pi fT)e^{-j2\pi ft_d} & |f| < B \ 0 & |f| > B \end{cases}
A pulse g(t) band-limited to B Hz is applied at the input of this filter. Find the output y(t).
This filter has ideal phase and nonideal magnitude characteristics.
Because g(t) \Leftrightarrow G(f), y(t) \Leftrightarrow Y(f) and
Y(f) = G(f)H(f) = G(f)\Pi(\frac{f}{2B})(1 + k \cos 2\pi fT)e^{-j2\pi ft_d}
Note that in the derivation of Eq. (3.62) because g(t) is band-limited to B Hz, we have
G(f) \cdot \Pi(\frac{f}{2B}) = G(f). Then, by using the time-shifting property and Eq. (3.32a), we have:
y(t) = g(t - td) + \frac{k}{2}[g(t - td - T) + g(t - t_d + T)].
The output is actually g(t) + (k/2)[g(t - T) + g(t + T)] delayed by td. It consists of g(t) and its echoes shifted by \pm td.
The dispersion of the pulse caused by its echoes is evident from Fig. 3.29c. Ideal amplitude but nonideal phase response of H(f) has a similar effect.
Consider a channel effect: y = f(g)
Where y(t) = a0 + a1g(t) + a2g^2(t) + a3g^3(t) + … + a_kg^k(t).
Note for g(t) with bandwidth B, g^k(t) has bandwidth kB.
Causes frequency dispersion
Interference between frequency channels
What does equation (12) represent?
Effect of using power-efficient amplifiers?
At specific time deltas and frequencies, and where two paths have similar gains
Constructive interference - same phases
Destructive interference - opposite phases
This effect is frequency-selective fading.
In a mobile comms environment:
Multipath changes with time
Causes time variations (semi-periodic and random) and attenuation of signal known as fading
Possible to reduce fading using automatic gain control (AGC)
What will be the effect of noise in AGC?
Parseval’s theorem in the frequency domain
Signal energy can now be determined from g(t) in the time domain or from G(f) in the frequency domain
Energy of signal composed of energies for all frequency components of g(t)
Energy spectral density defined as \Psi_g(f) = |G(f)|^2
And from the second equation in previous slide we have
Eg = \int{-\infty}^{\infty} \Psi_g(f) df.
Spectra of most signals → ∞
Practical signals have finite energy - hence spectrum → 0 as f → ∞
There exists some bandwidth B for which the energy content at f > B becomes negligible
EXAMPLE: Select a B such that the energy inside B is 95% of the signal energy (other percentages possible)
Suppression of components beyond b results in approximate signal \hat{g}(t)
Homework: Ex 3.17 and 3.18
Let g(t) be a baseband signal band-limited to B Hz. The amplitude-modulated signal \phi(t) is
\phi(t) = g(t) \cos 2\pi f_0t
and the spectrum (Fourier transform) of \phi(t) is
\Phi(f) = \frac{1}{2}[G(f + f0) + G(f - f0)]
The ESD of the modulated signal \phi(t) is |\Phi(f)|^2, that is,
\Psi{\phi}(f) = \frac{1}{4}|G(f + f0) + G(f - f_0)|^2
If f0 \geq B, then G(f + f0) and G(f - f_0) are nonoverlapping (see Fig. 3.35), and
\Psi{\phi}(f) = \frac{1}{4}[|G(f + f0)|^2 + |G(f - f0)|^2] = \frac{1}{4} \Psig(f+f0) + \frac{1}{4} \Psig(f-f_0)
The energy of a signal is equal to the area under the ESD, i.e.
E{\phi} = \frac{1}{2}Eg , f_0 \geq B
Autocorrelation given by
\psig(\tau) = \int{-\infty}^{\infty} g(t)g^*(t + \tau) dt
Note that \psig(\tau) = \psig(-\tau)
The autocorrelation of g(t) and the ESD are Fourier transform pairs:
\psig(\tau) \Leftrightarrow \Psig(f)
Therefore:
\Psig(f) = F{\psig(\tau)} = \int{-\infty}^{\infty} \psig(\tau)e^{-j2\pi f\tau} d\tau
\psig(\tau) = F^{-1}{\Psig(f)} = \int{-\infty}^{\infty} \Psig(f)e^{j2\pi f\tau} df
NOTE: Fourier transform performed in respect of \tau instead of t
If x(t) and y(t) are the input and the corresponding output of a linear time-invariant (LTI) system, then
Y(f) = H(f)X(f)
Therefore,
|Y(f)|^2 = |H(f)|^2 |X(f)|^2
\Psiy(f) = |H(f)|^2\Psix(f)
Thus, the output signal ESD is |H(f)|^2 times the input signal ESD.
NOTE: Signal power is time-averaged signal energy
Pg = \lim{T \to \infty} \frac{E{gt}}{T} = \lim{T \to \infty} [\int{-\infty}^{\infty} \frac{|GT(f)|^2}{T} df]
= \int{-\infty}^{\infty} \lim{T \to \infty} \frac{|G_T(f)|^2}{T} df
As T increases, the duration of g(t) increases, and so its energy E{GT} - therefore GT(f) increases with T
Convergence implies |G_T(f)|^2 must grow at the same rate as T, which means that we can change the order of integral and limit in last step.
Power spectral density and power:
Sg(f) = \lim{T \to \infty} \frac{|G_T(f)|^2}{T}
Pg = \int{-\infty}^{\infty} S_g(f) df
The time autocorrelation function of real signal g(t) is defined as
Rg(\tau) = \lim{T \to \infty} \frac{1}{T} \int_{-T/2}^{T/2} g(t)g(t - \tau) dt
As with autocorrelation, time autocorrelation is an even function of \tau, i.e. Rg(\tau) = Rg(-\tau)
Fourier transform pairs:
Rg(\tau) \Leftrightarrow \lim{T \to \infty} \frac{|GT(f)|^2}{T} = Sg(f)
Energy Signals | Power Signals | |
---|---|---|
Energy/Power | Eg = \int{-\infty}^{\infty} g^2(t) dt | Pg = \lim{T \to \infty} \frac{1}{T} \int{-T/2}^{T/2} g^2(t) dt = \lim{T \to \infty} \frac{E_{gT}}{T} |
Autocorrelation | \psig(\tau) = \int{-\infty}^{\infty} g(t)g(t+\tau) dt | Rg(\tau) = \lim{T \to \infty} \frac{1}{T} \int{-T/2}^{T/2} g(t)g(t+\tau) dt = \lim{T \to \infty} \frac{\psi_{gT}(\tau)}{T} |
Spectral Density | \Psi_g(f) = | G(f) |
Transform Relation | \psig(\tau) \leftrightarrow \Psig(f) | Rg(\tau) \leftrightarrow Sg(f) |
Energy/Power (Freq) | Eg = \int{-\infty}^{\infty} \Psi_g(f) df | Pg = \int{-\infty}^{\infty} S_g(f) df |
Why not just use the Fourier transform of the signal?
Fourier transform of power signal generally does not exist
Fourier transform not available for random signals (random noise)
Autocorrelation function and PSD exists for power signals and random signals
If g(t) and y(t) are the input and output of an LTI system with a transfer function H(f), then
Sy(f) = |H(f)|^2Sg(f)
A noise signal ni(t) with PSD S{n_i}(f) = K is applied at the input of an ideal differentiator
Determine the PSD and the power of the output noise signal n_o(t).
The transfer function of an ideal differentiator is H(f) = j2\pi f.
If the noise at the demodulator output is n_o(t), then from Eq. (3.91),
S{no}(f) = |H(f)|^2 S{ni}(f) = |j2\pi f|^2 K
The output PSD S{no}(f) is parabolic.
The output noise power N_o is the area under the output PSD. Therefore,
No = \int{-B}^{B} K(2\pi f)^2 df = 2K \int_{0}^{B} (2\pi f)^2 df = \frac{8\pi^2 B^3 K}{3}
Consider cosine amplitude modulated by signal g(t): \phi(t) = g(t) \cos 2\pi f_0t,
then the PSD S_{\phi}(f) of the modulated signal is:
S{\phi}(f) = \frac{1}{4}[Sg(f + f0) + Sg(f - f_0)],
See detailed derivation in Sec 7.8 of textbook
Modulation shifts PSD to \pm f_0
Power of \phi(t) is half the power of g(t), i.e. P{\phi} = \frac{1}{2}Pg f_0 \geq B
NOTE: Section 3.9 is self-study / revision
In amplitude-based modulation, the carrier phase remains constant, i.e. let \theta_c = 0 without loss of generality
Consider a message signal m(t) with bandwidth B, where m(t) \Leftrightarrow M(f)
then
m(t)\cos(2\pi fct) \Leftrightarrow \frac{1}{2} [M(f + fc) + M(f - f_c)]
Lower sidebands with bandwidth B where -fc < f < fc
Upper sidebands with bandwidth B where f > fc and f < -fc
fc > B, else the spectra will overlap, however practically fc >> B
Consider a DSB-SC signal, multiplied by a cosine with the same frequency and phase as carrier:
e(t) = m(t)\cos^2(\omega_ct)
= \frac{1}{2} [m(t) + m(t)\cos(2\omega_ct)].
The Fourier transform of e(t) is given by:
E(f) = \frac{1}{2}M(f) + \frac{1}{4}[M(f + 2fc) + M(f - 2fc)]
For a baseband signal m(t) = \cos \omegamt = \cos 2\pi fmt,
find the DSB-SC signal and sketch its spectrum. Identify the USB and LSB. Verify that the
DSB-SC modulated signal can be demodulated by the demodulator in Fig. 4.1e.
The case in this example is referred to as tone modulation because the modulating signal
is a pure sinusoid, or tone, \cos \omega_mt
To clarify the basic concepts of DSB-SC modulation, we shall work this problem in
the frequency domain as well as the time domain. In
the frequency domain approach, we work with the signal spectra. The spectrum of the
baseband signal m(t) = \cos \omega_mt is given by
\frac{1}{2} [\delta (f - fm) + \delta(f + fm)]
= \pi [\delta(\omega - \omegam) + \delta(\omega + \omegam)]
In the time domain approach, we work directly with signals in the time domain. For the baseband signal m(t) = \cos \omegamt, the DSB-SC signal \phi{DSB-SC}(t) is
\phi{DSB-SC}(t) = m(t) \cos \omegact
= \cos \omegamt \cos \omegact
= \frac{1}{2} [\cos(\omegac + \omegam)t + \cos(\omegac - \omegam)t]
We now verify that the modulated signal \phi{DSB-Sc}(t) = \cos \omegamt \cos \omegact, when applied to the input of the demodulator in Fig. 4.1e, yields the output proportional to the desired baseband signal \cos \omegamt.
The signal e(t) in Fig. 4.1e is given by
e(t) = \cos \omegamt \cos^2 \omegact
= \frac{1}{2} \cos \omegamt (1 + \cos 2\omegaet)
Multiplier modulators
Nonlinear modulators
Switching modulators
Obtained from variable gain amp where the gain parameter is controlled by m(t)
Initially, multiplication of two signals over wide dynamic range was challenging - not such a problem anymore
However, some other efficient amplitude modulation techniques are now presented
Nonlinear element:
y(t) = ax(t) + bx^2(t)
z(t) as in figure:
z(t) = y1(t) - y2(t) = [ax1(t) + bx1^2(t)] - [ax2(t) + bx2^2(t]]
Let x1(t) = \cos \omegact + m(t) and x2(t) = \cos \omegact - m(t), then
z(t) = 2am(t) + 4bm(t) \cos \omega_ct
Modulation can be achieved by multiplying m(t) with any periodic signal
Such a periodic signal \phi(t) can be expressed by a trigonometric Fourier series:
\phi(t) = \sum{n=0}^{\infty} Cn \cos n\omegact + \thetan
hence
m(t)\phi(t) = \sum{n=0}^{\infty} Cnm(t) \cos n\omegact + \thetan
Spectrum M(\omega) shifted to \pm \omegac, \pm 2\omegac, …, \pm n\omega_c
Consider the Fourier series expansion for a square wave:
w(t) = \frac{1}{2} + \frac{2}{\pi} (\cos \omegact - \frac{1}{3} \cos 3\omegact + \frac{1}{5} \cos 5\omega_ct - … )
The signal m(t)w(t) is then given by:
m(t)w(t) = \frac{1}{2}m(t) + \frac{2}{\pi} (m(t) \cos \omegact - \frac{1}{3}m(t) \cos 3\omegact + \frac{1}{5} m(t) \cos 5\omega_ct - … )
Spectrum M(\omega) shifted to \pm \omegac, \pm 3\omegac, \pm 5\omega_c, …
Difference between this and chopper modulator?
Consider case where m(t) is multiplied by a bi-polar rectangular pulse train
The Fourier series expansion of the rectangular pulse train:
w0(t) = \frac{4}{\pi} (\cos \omegact - \frac{1}{3} \cos 3\omegact + \frac{1}{5} \cos 5\omegact - … )
resulting in a modulated signal:
m(t)w(t) = \frac{4}{\pi} (m(t) \cos \omegact - \frac{1}{3} m(t) \cos 3\omegact + \frac{1}{5} m(t) \cos 5\omega_ct - … )
How is this modulator related to the chopper modulator?
Examples 4.2 and 4.3 for self-study
DSB-SC is simple to analyse, but not so simple to implement
DSB-SC needs coherent demodulation (carrier at receiver with same frequency and phase)
Unknown transmission path and motion of TX or RX can induce Doppler effect, introducing frequency and phase shifts
Local oscillator needs to reproduce cosine with the same frequency and phase, using only the received signal
This requires the implementation of a phase-locked loop in the receiver, which is more expensive
Amplitude modulation (AM) is a modulation method which allows for simplified receiver implementation
This is at the expense of a higher power (more expensive transmitter)
Makes sense in the case of a single transmitter and multiple receivers (radio)
Consider the AM signal:
\Phi{AM}(t) = A \cos \omegact + m(t) \cos \omega_ct
= [A + m(t)] \cos \omega_ct
This results in the Fourier transform pair:
\Phi{AM}(t) \Leftrightarrow \frac{1}{2} [M(f + fc) + M(f - fc)] + \frac{A}{2} [\delta(f + fc) + \delta(f - f_c)]
How does this compare to DSB-SC?
What does the spectrum look like?
AM signal: \Phi{AM}(t) = [A + m(t)] \cos \omegact
Two cases of interest:
[A + m(t)] \geq 0 \forall t, or A \geq |m(t)|
[A + m(t)] < 0 \forall t, or A < |m(t)|
If A \geq m(t), then we can recover m(t) by detecting the “envelope” of \Phi_{AM}(t)
Definition: envelope
If E(t) varies slowly in comparison with the sinusoidal carrier \cos \omegact, then the envelope of E(t) \cos \omegact is |E(t)|
Therefore, envelope detection is only possible if and only if |A + m(t)| = A + m(t)
To summarise,
f_c >> the bandwidth of m(t)
A + m(t) \geq 0, \forall t
m(t) with zero offset
Let \pm mp the max and min values of m(t), then the condition for envelope detection implies A \geq mp
Modulation index defined as
\mu = \frac{m_p}{A}
\mu = \frac{b}{A}
For distortionless envelope detection:
0 \leq \mu \leq 1
m(t) with nonzero offset
In some occasions m(t) can have a nonzero offset, i.e. m{min} ≠ m{max}
Still, in this case, for distortionless envelope detection:
0 \leq \mu \leq 1,
but here
\mu = \frac{m{max} - m{min}}{2A + m{max} + m{min}}
Sketch \phi{AM}(t) for modulation indices of \mu = 0.5 and \mu = 1 when m(t) = b \cos \omegamt.
This case is referred to as tone modulation because the modulating signal is a pure sinusoid (or tone).
In this case, m{max} = b and m{min} = -b.
Hence the modulation index according to Eq. M = \frac{b}{A}
therefore,
m(t) = b \cos \omegamt = \mu A \cos \omegamt
\phi{AM} (t) = [A + m(t)] \cos \omegaet = A[1 + \mu \cos \omegamt] \cos \omegaet
The advantage of AM envelope detection comes at a cost
The carrier term does not carry information, and the carrier power is wasteful:
\Phi{AM}(t) = A \cos \omegact \overset{\text{carrier}}{} + m(t) \cos \omega_ct \overset{\text{sidebands}}{}
The power of A \cos \omegact and m(t) \cos \omegact are [see textbook eq 3.93]:
P_c = \frac{A^2}{2}
Ps = \frac{1}{2}Pm
The power efficiency is given by
\eta = \frac{\text{useful power}}{\text{total power}} = \frac{Ps}{Ps + Pc} = \frac{Pm}{P_m + \frac{A^2}{2}}
For the special case of tone modulation:
\eta_{max} = 33\%
Determine \eta and the percentage of the total power carried by the sidebands of the AM wave for tone modulation when \mu = 0.5 and when \mu = 0.3.
For \mu = 0.5
\eta = \frac{\mu^2}{2 + \mu^2} 100\% = \frac{(0.5)^2}{2 + (0.5)^2} 100\% = 11.11\%
Hence, only about 11% of the total power is in the sidebands. For \mu = 0.3
\eta = \frac{\mu^2}{2 + \mu^2} 100\% = \frac{(0.3)^2}{2 + (0.3)^2} 100\% = 4.3\%
Hence, only 4.3% of the total power is in the sidebands that contain the message signal.
Modulation
Same as DSB-SC, but add a carrier component A \cos \omega_ct
Demodulation
Rectifier detector
Envelope detector
In essence, the signal is multiplied by a periodic signal w(t) (approximated by a rectangular pulse train), resulting in the rectifier output:
vR(t) = { [A + m(t)] \cos \omegact } w(t)
= [A + m(t)] \cos \omegact [ \frac{1}{2} + \frac{2}{\pi} (\cos \omegact - \frac{1}{3} \cos 3\omegact + \frac{1}{5} \cos 5\omegact - … )]
= \frac{1}{\pi} [A + m(t)] + \text{other higher frequency terms}
After the low pass filter with cutoff B Hz, the output is [A + m(t)]/\pi
Important design constraint of RC circuit
\frac{1}{\omega_c} << RC < \frac{1}{2\pi B}
\Rightarrow 2\pi B < \frac{1}{RC