Fourier series is powerful for periodic signals, but most real-world signals are non-periodic. We need a way to analyze these non-periodic signals in the frequency domain.
B. The Basic Idea
A non-periodic signal can be viewed as a periodic signal with an infinite period (Tโโ).
As T increases for a sequence of periodic signals (each replicating a finite-duration aperiodic signal x(t) within its period), the periodic signals converge to the non-periodic signal x(t).
Correspondingly, their spectra (which are line spectra) should converge to the spectrum of the non-periodic signal. As Tโโ, the fundamental frequency ฯ0โ=2ฯ/Tโ0, meaning the lines in the spectrum become infinitely close, forming a continuous spectrum.
C. Deriving the Fourier Transform
Step 1: Construct a periodic signal x~Tโ(t)
Let x(t) be a non-periodic signal such that x(t)=0 for โฃtโฃ>T1โ.
Construct a periodic signal x~Tโ(t) with period T>2T1โ, such that x~Tโ(t)=x(t) for โฃtโฃ<T/2.
Then, limTโโโx~Tโ(t)=x(t).
Since x~Tโ(t) is periodic (with ฯ0โ=2ฯ/T), its Fourier series coefficients are:
Letโs define X(jฯ)โโซโโโโx(t)eโjฯtdt as the Fourier Transform of x(t).
Then, Takโ=X(jฯ)โฃฯ=kฯ0โโ.
The quantity Takโ represents samples of the envelope X(jฯ). As Tโโ, ฯ0โโ0, and the discrete samples kฯ0โ become a continuous variable ฯ.
Thus, as Tโโ, TakโโX(jฯ), which is the spectrum of the non-periodic signal x(t).
The analysis equation of Fourier Transform is:
X(jฯ)=โซโโโโx(t)eโjฯtdt
Step 3: Derive the Fourier Transform Synthesis Equation
This sum can be seen as an approximation of an integral. Each term X(jkฯ0โ)ejkฯ0โtฯ0โ represents the area of a rectangle of height X(jkฯ0โ)ejkฯ0โt and width ฯ0โ.
Actually this fits the definition method of Riemann Integral.
As Tโโ, ฯ0โโ0, and x~Tโ(t)โx(t). The sum becomes an integral:
This is the synthesis equation (Inverse Fourier Transform).
Summary of Derivation:
Given a non-periodic x(t) (compactly supported, though this can be relaxed in formal proof), construct periodic x~Tโ(t) such that limTโโโx~Tโ(t)=x(t).
Show that as Tโโ, TakโโX(jฯ)โโซโโโโx(t)eโjฯtdt. Define X(jฯ) as the spectrum of x(t).
Show that as Tโโ, โakโejkฯ0โt converges to 2ฯ1โโซX(jฯ)ejฯtdฯ. Thus, x(t)=2ฯ1โโซX(jฯ)ejฯtdฯ is the synthesis equation.
V. Fourier and Inverse Fourier Transforms ๐
The Fourier Transform pair is defined as:
Fourier Transform (Analysis Equation):
X(jฯ)=F{x(t)}=โซโโโโx(t)eโjฯtdt
This transforms the signal x(t) into its spectrum X(jฯ).
This synthesizes the signal x(t) from its spectrum X(jฯ).
There is a notable symmetry between the transform and its inverse.
Comparison with Fourier Series:
Synthesis Equations:
Periodic: xT0โโ(t)=โk=โโ+โโakโejkฯ0โt (linear combination of harmonically related complex exponentials (HRCEs) with fundamental frequency ฯ0โ).
Non-periodic: x(t)=2ฯ1โโซโโโโX(jฯ)ejฯtdฯ (linear combination of HRCEs with an infinitely small fundamental frequency).
Analysis Equations:
Periodic: akโ=T0โ1โโซT0โโxT0โโ(t)eโjkฯ0โtdt (generates a discrete spectrum).
Non-periodic: X(jฯ)=โซโโโโx(t)eโjฯtdt (generates a continuous spectrum as โeffective fundamental frequencyโ is dฯโ0).
Example 2: x(t)=eโaโฃtโฃ, for a>0 X(jฯ)=โซโโ0โeateโjฯtdt+โซ0โโeโateโjฯtdt X(jฯ)=โซ0โโeโatejฯtdt+โซ0โโeโateโjฯtdt (by changing variable tโโt in the first integral) X(jฯ)=โซ0โโeโ(aโjฯ)tdt+โซ0โโeโ(a+jฯ)tdt X(jฯ)=aโjฯ1โ+a+jฯ1โ=(aโjฯ)(a+jฯ)(a+jฯ)+(aโjฯ)โ=a2+ฯ22aโ.
Magnitude: โฃX(jฯ)โฃ=X(jฯ) (since itโs real and positive).
Phase: โ X(jฯ)=0.
Example 3: x(t)=ฮด(t) (Dirac delta function) X(jฯ)=โซโโโโฮด(t)eโjฯtdt=eโjฯโ 0=1 (by the sifting property of the delta function).
The spectrum is constant, meaning ฮด(t) is composed of all frequencies with equal strength (amplitude 1 and phase 0).
VII. Convergence of the Fourier Transform ๐
The convergence criteria for Fourier Transform are inherited from those for Fourier Series, as FT is effectively FS when Tโโ.
Two types of convergence:
Finite Energy Condition: If x(t) has finite energy, i.e., โซโโโโโฃx(t)โฃ2dt<โ, then the Fourier Transform converges in the sense that the error โซโโโโโฃx(t)โx^(t)โฃ2dt=0, where x^(t)=Fโ1{F{x(t)}}. Such signals are often bounded and decay to 0 sufficiently fast as โฃtโฃโโ.
Dirichlet Conditions: If x(t) satisfies the Dirichlet conditions, x^(t) converges pointwise to x(t) where x(t) is continuous, and to the average of the values on either side of a discontinuity.
The Dirichlet conditions are:
a. x(t) is absolutely integrable over the entire time domain: โซโโโโโฃx(t)โฃdt<โ. ็ปๅฏนๅฏ็งฏ
b. x(t) has a finite number of maxima and minima within any finite interval. ๆ้ไธชๆๅผ็น
c. x(t) has a finite number of discontinuities, each with a finite height, within any finite interval. ๆ้ไธช้ดๆญ็น๏ผไธ้ฝๆฏ็ฌฌไธ็ฑป้ดๆญ็น
For x(t)=eโatu(t),a>0: Satisfies Dirichlet conditions.
For x(t)=eโaโฃtโฃ,a>0: Satisfies Dirichlet conditions.
For x(t)=ฮด(t): Violates the third Dirichlet condition (infinite discontinuity). However, its FT is still considered to converge because Dirichlet conditions are sufficient, not necessary.
VIII. Rectangle-Sinc Fourier Transform Pair
If X(jฯ)=F{x(t)}, then x(t) and X(jฯ) are a Fourier transform pair. The rectangle-sinc pair is very important.
Example 4.5: Rectangular Pulse in Frequency Domain (Ideal Lowpass Filter Spectrum)
Let X(jฯ)={1,0,โโฃฯโฃ<Wโฃฯโฃ>Wโ
We use the inverse Fourier transform to find x(t):
A rectangular signal in time, x(t), has a sinc-shaped spectrum, X(jฯ).
A rectangular spectrum in frequency, X(jฯ), corresponds to a sinc-shaped signal in time, x(t).
Duality Property: If a signal x(t) has a spectrum X(jฯ) (i.e., x(t)โX(jฯ)), then a signal in time that has the functional form of X(t) will have a spectrum 2ฯx(โฯ) (i.e., X(t)โ2ฯx(โฯ)). This will be formally proven in the next lecture.
Lecture 9: Fourier Transform Properties
FT and FS (Fourier Series)
The Fourier Series (FS) represents a periodic signal ฯT0โโ(t) as a sum of weighted harmonics, while the Fourier Transform (FT) represents a signal x(t) in the continuous frequency domain.
The Fourier series spectrum (akโ) is a discrete-time signal, leading to a sum in the synthesis equation.
The Fourier transform spectrum (X(jฯ)) is a continuous-time signal, leading to an integral in the synthesis equation.
These differences arise because a non-periodic signal can be seen as having an infinite period, so its representation includes harmonics at all frequencies.
Normalized sinc Function
The normalized sinc function is defined as:
sinc(ฯ)=ฯฯsin(ฯฯ)โ
An arbitrary rectangular pulse x(t) which is 1 for โฃtโฃ<T1โ and 0 otherwise, has a Fourier Transform:
X(jฯ)=ฯ2sin(ฯT1โ)โ
This can be rewritten using the normalized sinc function:
X(jฯ)=2T1โsinc(ฯT1โโฯ)
Fourier Transform for Periodic Signals
To find the FT spectrum of a periodic signal xT0โโ(t), we look for X(jฯ) such that:
xT0โโ(t)=2ฯ1โโซโโโโX(jฯ)ejฯtdฯ
If akโ are the FS coefficients of xT0โโ(t), then we must have:
This is the Fourier series synthesis equation, confirming the result.
Convergence Note: Periodic signals donโt satisfy the Dirichlet conditions or finite energy criterion for FT convergence. However, convergence is still guaranteed here because these conditions are sufficient, not necessary.
Examples of FT for Periodic Signals
x(t)=cos(ฯ0โt)
FS coefficients: a1โ=1/2, aโ1โ=1/2, and akโ=0 for other k.
FT: X(jฯ)=ฯฮด(ฯโฯ0โ)+ฯฮด(ฯ+ฯ0โ)
Difference between FT and FS spectra: The FT spectrum consists of impulses (not just numbers), is rescaled by 2ฯ, and the horizontal axis represents ฯ instead of k.
x(t)=sin(ฯ0โT)
FS coefficients: a1โ=2j1โ, aโ1โ=โ2j1โ, and akโ=0 for other k.
If x(t)โX(jฯ) and y(t)โY(jฯ), then ax(t)+by(t)โaX(jฯ)+bY(jฯ) for any complex numbers a and b.
This is true because integration is a linear operation.
Time-Shifting Property: If x(t)โX(jฯ), then x(tโt0โ)โeโjฯt0โX(jฯ)
For FS: x(tโt0โ)โทF.S.โeโjkฯ0โt0โakโ
Indication: Time shifting only changes the phase spectrum by adding a linear phase โฯt0โ. The phase spectrum determines the location of each harmonic.
Time-Reversal: If x(t)โX(jฯ), then x(โt)โX(โjฯ)
For FS: x(โt)โaโkโ
Indications:
If x(t) is even, then X(jฯ) is even.
If x(t) is odd, then X(jฯ) is odd.
Reversing a signal or flipping an image causes a reversed or flipped spectrum.
Conjugacy and Symmetry
If x(t)โX(jฯ), then xโ(t)โXโ(โjฯ)
For FS: xโ(t)โaโkโโ
If x(t) is real, then X(jฯ)=Xโ(โjฯ) (conjugate symmetric).
This means the real part of X(jฯ) is even, and the imaginary part of X(jฯ) is odd.
For real signals x(t):
If x(t) is real and even, X(jฯ) is real and even.
If x(t) is real and odd, X(jฯ) is imaginary and odd.
Since a real signal x(t)=xeโ(t)+xoโ(t) (even and odd components), its FT is X(jฯ)=F{xeโ(t)}+F{xoโ(t)}.
Also, X(jฯ)=Re{X(jฯ)}+jโ Im{X(jฯ)}.
This implies: Re{X(jฯ)}=F{xeโ(t)} and jIm{X(jฯ)}=F{xoโ(t)}.
Example 4.9 (Combining signals):
How to generate Fourier transform of x(t) from that of x1(t) & x2(t)?
Given x1โ(t) with X1โ(jฯ)=ฯ2sin(ฯ/2)โ and x2โ(t) with X2โ(jฯ)=ฯ2sin(3ฯ/2)โ.
Find that x(t)=21โx1โ(tโ2.5)+x2โ(tโ2.5).
Then X(jฯ)=eโj2.5ฯ(21โX1โ(jฯ)+X2โ(jฯ))=eโj2.5ฯ(ฯsin(ฯ/2)โ+ฯ2sin(3ฯ/2)โ).
Example 4.10 (Using properties to find FT): Find FT of x(t)=eโaโฃtโฃ, a>0, given eโatu(t)โa+jฯ1โ.
Decompose x(t) into x(t)=eโatu(t)+eatu(โt).
Let y(t)=eโatu(t). Then eatu(โt)=y(โt)
Then, we have Y(jฯ)=a+jฯ1โ, and F{y(โt)}=Y(โjฯ)=aโjฯ1โ.
So, X(jฯ)=a+jฯ1โ+aโjฯ1โ=(a+jฯ)(aโjฯ)aโjฯ+a+jฯโ=a2+ฯ22aโ.
Parsevalโs Theorem
If x(t)โX(jฯ), then โซโโโโโฃx(t)โฃ2dt=2ฯ1โโซโโโโโฃX(jฯ)โฃ2dฯ
This is also called the โenergy-preservation propertyโ.
โฃX(jฯ)โฃ2 is called the Energy-density Spectrum๏ผๅ็ๅฏๅบฆ่ฐฑ๏ผ็ฎ็งฐๅ็่ฐฑ๏ผ of x(t), reflecting energy density at each frequency.
For natural signals, energy is typically more concentrated in lower frequencies.
Time-Scaling Property
If x(t)โFX(jฯ), then for any real a๎ =0: x(at)โFโฃaโฃ1โX(ajฯโ)
If โฃaโฃ>1, x(t) is squeezed, X(jฯ) is stretched and scaled down.
If โฃaโฃ<1, x(t) is stretched, X(jฯ) is squeezed and scaled up.
Squeezing in time (faster variation) leads to more high-frequency components (stretching in frequency).
Stretching in time (slower variation) leads to more low-frequency components (squeezing in frequency).
Example: rect(t/T1โ)โ2โฃT1โโฃsinc(ฯT1โฯโ), where rect(t)=1 for โฃtโฃ<1 and 0 otherwise. (Note: The slide shows rect(t)โ2sinc(ฯฯโ) with T1โ=1 for the basic form).
Proof of Time-Scaling Property: F{x(at)}=โซโโโโx(at)eโjฯtdt.
Let u=at, so t=u/a and dt=du/a.
If a>0: โซโโโโx(u)eโjฯ(u/a)aduโ=a1โโซโโโโx(u)eโj(ฯ/a)udu=a1โX(jaฯโ).
If a<0: The limits of integration flip, โซโโโโ, which introduces a negative sign that cancels with 1/a when taking โฃaโฃ. โซโโโโx(u)eโjฯ(u/a)aduโ=โa1โโซโโโโx(u)eโj(ฯ/a)udu=โฃaโฃ1โX(jaฯโ).
Thus, for a๎ =0, F{x(at)}=โฃaโฃ1โX(jaฯโ).
Duality and Related Properties
The FT and IFT equations are symmetric, differing mainly by a sign in the exponent and a 1/(2ฯ) scaler.
Derived from differentiation property and duality. Proof (using duality twice):
Differentiation: dtdy(t)โโjฯY(jฯ).
Duality: X(t)โ2ฯx(โjฯ).
Apply differentiation to X(t): dX(t)/dtโjฯ(2ฯx(โjฯ)). (Here, the FT is taken with respect to ฯ).
Apply duality again to the pair from step 3:
The โtime functionโ is jฯโฒ(2ฯx(โjฯโฒ)) and the โfrequency functionโ is dX(t)/dt.
Using Xโฒ(tโฒ)โ2ฯxโฒ(โjฯโฒ): j(โjt)(2ฯx(โj(โjt)))โ2ฯd(โฯ)d(X(โฯ))โ/(2ฯ) ? This is also tricky to follow directly.
A more standard derivation: ๆดๆ ๅ็ไฝฟ็จๆนๆณไป็ถๆฏไฝฟ็จ็งฏๅ๏ผไธ่ฆไฝๆญป็ปๆฅ็ปๅป๏ผ
The term 2ฯ1โโซโโโโejฯ(tโt1โโt2โ)dฯ is the inverse Fourier transform of 1โ ejฯ(tโt1โโt2โ), which is ฮด(tโt1โโt2โ).
Using the sifting property of the Dirac delta function: โซโโโโy(t2โ)ฮด(tโt1โโt2โ)dt2โ=y(tโt1โ).
=โซโโโโx(t1โ)y(tโt1โ)dt1โ
This is the definition of the convolution x(t)โy(t).
Output Prediction for LTI Systems
For an LTI system with impulse response h(t) and an arbitrary input signal x(t), the output y(t) is x(t)โh(t)=y(t).
Based on the convolution property:
Y(jฯ)=X(jฯ)H(jฯ)
where H(jฯ) is the systemโs frequency response.
Thus, the output spectrum of an LTI system equals the input spectrum multiplied by the frequency response. This extends the frequency-domain approach to almost any input signal.
Examples of Predicting Output in the Frequency Domain
Example 4.15: Delayer System
For a delayer system y(t)=x(tโt0โ):
The impulse response is h(t)=ฮด(tโt0โ).
The frequency response is H(ฯ)=eโjฯt0โ.
The output spectrum is Y(ฯ)=eโjฯt0โX(ฯ), which is consistent with F{x(tโt0โ)}.
Example 4.16: Differentiator System
For a differentiator system y(t)=dx(t)/dt:
The impulse response is h(t)=dtdฮด(t)โ.
The frequency response is H(ฯ)=jฯโ 1=jฯ.
The output spectrum is Y(ฯ)=jฯX(ฯ), consistent with F{dx(t)/dt}.
Example 4.19: Convolution using Frequency Domain
Given h(t)=eโatu(t) for a>0 and x(t)=eโbtu(t) for b>0, with a๎ =b. Determine y(t) using the convolution property.
X(jฯ)=b+jฯ1โ and H(jฯ)=a+jฯ1โ.
Y(jฯ)=X(jฯ)H(jฯ)=(a+jฯ)(b+jฯ)1โ.
Using partial-fraction expansion:
Y(jฯ)=a+jฯAโ+b+jฯBโ
where A=bโa1โ and B=โbโa1โ.
So, Y(jฯ)=bโa1โ[a+jฯ1โโb+jฯ1โ].
Taking the inverse Fourier transform (IFT), using the pair eโctu(t)โc+jฯ1โ:
y(t)=bโa1โ[eโatu(t)โeโbtu(t)]
Convolution with a Periodic Impulse Train
If the impulse response is a periodic impulse train, say p(t)=โk=โโโโฮด(tโkT0โ), its Fourier transform is P(jฯ)=T0โ2ฯโโk=โโโโฮด(ฯโkฯ0โ), where ฯ0โ=2ฯ/T0โ.
If an input x(t) with spectrum X(jฯ) is convolved with such a train, the output spectrum Y(jฯ) becomes a sampled version of X(jฯ).
Proof of F{xT0โโ(t)}=2ฯโk=โโโโakโฮด(ฯโkฯ0โ)
Let xT0โโ(t) be a periodic signal formed by replicating a signal x~(t) every T0โ. This can be expressed as xT0โโ(t)=x~(t)โโk=โโโโฮด(tโkT0โ).
Using the convolution property:
Recognizing that the Fourier series coefficients akโ=T0โX~(kฯ0โ)โ (for one period of x~(t) defined from โT0โ/2 to T0โ/2 or similar, and X~(ฯ) is the FT of one period):
=2ฯk=โโโโโakโฮด(ฯโkฯ0โ)
FT Spectrum of Periodic Signals: Example - Periodic Square Wave
Let x~(t) be a rectangular pulse: x~(t)={1,0,โโฃtโฃ<T1โโฃtโฃ>T1โโ.
Its Fourier transform is X~(jฯ)=ฯ2sin(ฯT1โ)โ=2T1โsinc(ฯT1โฯโ).
A periodic square wave x(t) can be seen as the convolution of x~(t) with an impulse train โk=โโโโฮด(tโkT0โ).
Using the convolution property, the FT of the periodic square signal is:
Consider a system with x(t) as input. The error signal e(t) is x(t)+(โ3y(t))=x(t)โ3y(t). The block D is a differentiator, so y(t)=dtde(t)โ.
In the frequency domain:
Rearranging for the frequency response H(jฯ)=Y(jฯ)/X(jฯ) (this specifically gives the frequency response of the LTI subsystem for the zero-state response):
This is the general expression for the frequency response of a stable LCCDE systemโs LTI part.
Example 4.25: Impulse Response of a 2nd-order LCCDE System
Given the LCCDE: dt2d2y(t)โ+4dtdy(t)โ+3y(t)=dtdx(t)โ+2x(t). Assume the system is stable.
Taking the Fourier transform:
A(jฯ+3)+B(jฯ+1)=jฯ+2.
If jฯ=โ1, A(โ1+3)=โ1+2โ2A=1โA=1/2.
If jฯ=โ3, B(โ3+1)=โ3+2โโ2B=โ1โB=1/2.
So,
H(jฯ)=jฯ+121โโ+jฯ+321โโ
The impulse response (for the stable system) is:
h(t)=21โeโtu(t)+21โeโ3tu(t)
Example 4.26: Output for a given input
For the same system, find the output y(t) for x(t)=eโtu(t). X(jฯ)=jฯ+11โ. Y(jฯ)=H(jฯ)X(jฯ)=[(jฯ+1)(jฯ+3)jฯ+2โ][jฯ+11โ]=(jฯ+1)2(jฯ+3)jฯ+2โ.
Using partial-fraction expansion for repeated roots:
To use this Fourier transform method for LCCDEs, we usually require the system to be stable.
Stability (BIBO) implies h(t) is absolutely integrable, which is a Dirichlet condition for the convergence of H(jฯ).
Zero-State Response:
This frequency response method directly predicts the zero-state response.
If initial conditions are zero, or the system is LTI (implying initial rest for causality), then the full response is the zero-state response. Otherwise, the zero-input response must be solved separately (e.g., using the characteristic polynomial method).
Exercise: RC Circuit (Analog Lowpass Filter)
Consider an RC circuit with input voltage Vinโ (or vsโ(t)) across a series resistor R and capacitor C. The output voltage Voutโ (or vcโ(t)) is across the capacitor.
LCCDE: RCdtdvcโ(t)โ+vcโ(t)=vsโ(t), which can be written as dtdvcโ(t)โ+RC1โvcโ(t)=RC1โvsโ(t).
Frequency Response:
Taking FT: RCjฯVcโ(jฯ)+Vcโ(jฯ)=Vsโ(jฯ). Vcโ(jฯ)(RCjฯ+1)=Vsโ(jฯ).
H(jฯ)=Vsโ(jฯ)Vcโ(jฯ)โ=1+RCjฯ1โ
Magnitude Spectrum:
โฃH(jฯ)โฃ=1+(RCฯ)2โ1โ
This system acts as an analog first-order lowpass filter. It allows low-frequency signals to pass while attenuating high-frequency signals.
Multiplication Property
The multiplication property is dual to the convolution property.
If s(t)โS(jฯ) and p(t)โP(jฯ), then:
Multiplication in the time domain corresponds to convolution in the frequency domain (scaled by 1/(2ฯ)).
Proof (using duality)
Given s(t)โS(ฯ) and p(t)โP(ฯ).
By duality, S(t)โ2ฯs(โฯ) and P(t)โ2ฯp(โฯ).
Using the convolution property: S(t)โP(t)โ(2ฯs(โฯ))(2ฯp(โฯ))=4ฯ2s(โฯ)p(โฯ).
Let g(t)=S(t)โP(t) and G(ฯ)=4ฯ2s(โฯ)p(โฯ).
Applying duality again to g(t)โG(ฯ): G(t)โ2ฯg(โฯ). 4ฯ2s(โt)p(โt)โ2ฯ[S(โฯ)โP(โฯ)].
Replacing t with โt: 4ฯ2s(t)p(t)โ2ฯ[S(ฯ)โP(ฯ)] (assuming S and P are transforms from tโฯ, so the convolution arguments become ฯ).
Therefore, s(t)p(t)โ2ฯ1โ[S(jฯ)โP(jฯ)].
Example 4.23: FT of a product of sinc functions
Find the Fourier transform of
x(t)=ฯt2sin(t)sin(t/2)โ
Rewrite x(t) as:
x(t)=ฯ(ฯtsin(t)โ)(ฯtsin(t/2)โ)
Let s1โ(t)=ฯtsin(t)โ and s2โ(t)=ฯtsin(t/2)โ.
This is the convolution of two rectangular functions. S1โ(jฯ) is a rectangle from ฯ=โ1 to 1 with height 1. S2โ(jฯ) is a rectangle from ฯ=โ1/2 to 1/2 with height 1.
The convolution of these two rectangular pulses will result in a trapezoidal๏ผๆขฏๅฝข๏ผ pulse.
The convolution will range from (โ1โ1/2) to (1+1/2), i.e., from โ3/2 to 3/2.
The flat top will be from (โ1+1/2) to (1โ1/2), i.e., from โ1/2 to 1/2. The height of the convolution is Area(S2โ(jฯ)) = 1ร1=1.
So X(jฯ) will be a trapezoid with:
Value 0 for ฯ<โ3/2 and ฯ>3/2.
Linearly increasing from 0 at ฯ=โ3/2 to 1/2ร1=1/2 at ฯ=โ1/2.
Constant value 1/2 for โ1/2โคฯโค1/2.
Linearly decreasing from 1/2 at ฯ=1/2 to 0 at ฯ=3/2.
Modulation and Demodulation
Modulation System (Example 4.21)
Assume a bandlimited signal s(t) with spectrum S(jฯ) (bandlimited to ฯ1โ, i.e., S(jฯ)=0 for โฃฯโฃ>ฯ1โ).
Multiply s(t) with a high-frequency sinusoidal function p(t)=cos(ฯ0โt), where ฯ0โโซฯ1โ.
The Fourier transform of p(t)=cos(ฯ0โt) is P(jฯ)=ฯ[ฮด(ฯโฯ0โ)+ฮด(ฯ+ฯ0โ)].
The resulting signal is r(t)=s(t)p(t).
Its spectrum R(jฯ) is given by the multiplication property:
Modulation is used in wireless transmission to send multiple signals over the same channel by modulating them onto different carrier frequencies.
Transmitter Side: Multiple signals (e.g., voices) are modulated using different carrier frequencies (e.g., cos(2ฯf1โt), cos(2ฯf2โt), etc.). The modulated signals are then added together. In the frequency domain, their spectra are shifted to different frequency bands and do not overlap.
Receiver Side: To recover a specific signal (e.g., signal 1):
Multiply the combined received signal by the corresponding carrier frequency (e.g., cos(2ฯf1โt)). This shifts the desired signalโs spectrum back to baseband (around ฯ=0) and other signals to other frequencies.
Apply a lowpass filter to extract the baseband signal, recovering the original signal.
This technique is known as Amplitude Modulation (AM).
Generating a Bandpass Filter from a Lowpass Filter
A bandpass filter can be created from an ideal lowpass filter HLPโ(jฯ) (with cutoff ฯcโ) using modulation.
Let the input signal be x(t) and its spectrum X(jฯ).
Modulate the input signal: x(t)ejฯcโt. Its spectrum is X(j(ฯโฯcโ)).
Pass this through the lowpass filter HLPโ(jฯ). The output spectrum is HLPโ(jฯ)X(j(ฯโฯcโ)). This is a version of X(jฯ) that was originally around ฯcโ, now shifted to baseband and filtered.