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Communication Systems Engineering Episode 1 Part 3 docx

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• When sampling at rate 2W the reconstruction filter must be a rectangular pulse – Such a filter is not realizable – For perfect reconstruction must look at samples in the infinite fut

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Given a continuous time waveform, can we represent it using discrete samples?

– How often should we sample?

– Can we reproduce the original waveform?

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Frequency representation of signals

( ) = ∫−∞ x(t)e

j ft

dt

2 π

x t( ) = ∫−∞ X( f )e

j ft

df

Notation:

X(f) = F[x(t)]

X(t) = F-1 [X(f)]

x(t) X(f)

Eytan Modiano

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Unit impulse δ (t)

t 0,

δ ( ) = ∀t ≠ 0

δ ( ) = 1

∫−∞ t

δ ( ) ( ) =

∫−∞ t x t

∫δ(t − τ)x(τ ) =

F[ (t)] =∫−∞ δ(t)ej ft dt = e0 = 1 δ ( )

t

δ ( ) ⇔ 1

0

1

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j t j f

Rectangle pulse

t

 1 | | < 1 / 2

Π ( ) = 1 2 | t |= 1 2

 0 otherwise

1 2

F[ (t)] =∫−∞ Π(t)ej ft

dt = ∫− 1 2

ej 2πft

dt

/

e− π − e π πf

= = Sin( ) = Sinc f

j2 π f πf ( )

Π ( )t

1

1/2

Eytan Modiano

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ααα βββ ααα βββ

τττ πππ τττ

Properties of the Fourier transform

Linearity

– x1(t) <=> X1(f), x2(t) <=>X2(f) => αx1(t) + βx2(t) <=> αX1(f) + βX2(f)

Duality

– X(f) <=> x(t) => x(f) <=> X(-t) and x(-f)<=> X(t)

Time-shifting: x(t-τ) <=> X(f)e -j2πfτ

Scaling: F[(x(at)] = 1/|a| X(f/a)

Convolution: x(t) <=> X(f), y(t) <=> Y(f) then,

– F[x(t)*y(t)] = X(f)Y(f) – Convolution in time corresponds to multiplication in frequency and

visa versa

x t ( ) * y(t) = ∫−∞x t ( − τ)y(τ)d

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Fourier transform properties (Modulation)

2 π

x t e( ) j f o t X( f f o )

e jx + ejx

Now, cos(x) =

2

x t e j f o t + x t e j 2πf o t

x t( ) cos(2 πf o t) = ( ) 2π ( )

2

(

x t Hence, ( ) cos(2 πf o t) X f f o ) + X( f + f o )

2

Example: x(t)= sinc(t), F[sinc(t)] = Π(f)

Y(t) = sinc(t)cos(2πf o t) <=> (Π(f-f o )+Π(f+f o ))/2

1/2

o

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| (

Power content of signal

Autocorrelation

∫∞ x t

−∞| ( ) |

2

dt =∫−∞ X f ) |

2

df

*

R x ( ) τ = ∫−∞ x(t)x (t − τ)dt

R x ( ) τ ⇔ | X( f ) |2

x t( ) o = x t ( ) δ(t t o )

x t( ) ∑δ(t nt o ) = sampled version of x(t)

n =−∞

F[ ∑ δ(t nt o )] = 1 ∑ δ( f n )]

n =−∞ o n =−∞ o

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The Sampling Theorem

X f( ) ( ) = 0, for all f ,| f | W

– Bandwidth < W

Sampling Theorem: If we sample the signal at intervals Ts where

Ts <= 1/ 2W then signal can be completely reconstructed from its samples using the formula

x t( ) = ∑2W©T s x(nT s ) sin c[2W©(t nT s )]

n =−∞

Where, W W©≤ 1 − W

T s

With T s = => x t( ) = ∑ x(nT s ) sin c[( n)]

n =−∞ s

( ) = ∑ x( ) sin [2

n =−∞

Eytan Modiano

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Proof

x t δ( ) = x t ( ) ∑δ(t nT s )

n =−∞

Xδ( ) f = X f ( ) * F[ ∑δ(t nT s )]

n =−∞

F[ ∑δ(t nT s )] = 1 ∑ δ( f n )

n =−∞ T s n =−∞ T s

δ( ) = ∑ X f − )

Ts n =−∞ T s

The Fourier transform of the sampled signal is a replication of the Fourier transform of the original separated by 1/Ts intervals

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Proof, continued

If 1/Ts > 2W then the replicas of X(f) will not overlap and can be recovered

How can we reconstruct the original signal?

– Low pass filter the sampled signal

f

Ideal low pass filter is a rectangular pulse H f T( ) = Πs ( )

2W

Now the recovered signal after low pass filtering

f

X f ( ) = Xδ( ) f T sΠ( )

2W

f

x t( ) = F−1[ Xδ( f )T sΠ( )]

2W

( ) = ∑ x nT s )Sinc( n)

n =−∞ T s

Eytan Modiano

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When sampling at rate 2W the reconstruction filter must be a

rectangular pulse

– Such a filter is not realizable – For perfect reconstruction must look at samples in the infinite future

and past

In practice we can sample at a rate somewhat greater than 2W which makes reconstruction filters that are easier to realize

Given any set of arbitrary sample points that are 1/2W apart, can construct a continuous time signal band-limited to W

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