ee421, fall 1998 michigan technological university timothy j. schulz 08-sept, 98ee421, lecture 11...

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08-Sept, 98 EE421, Lecture 1 1 EE421, Fall 1998 Michigan Technological University Timothy J. Schulz Digital Signal Processing (DSP) Systems Digital processing of analog signals (mixed signal applications) forms one of the most important applications of DSP theory. A/D converter DSP system D/A converter ...101011... …001010... analog input digital input analog output digital output antialiasing prefilter sampling and quantization discrete to continuous reconstruction filter

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Page 1: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 1

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Digital Signal Processing (DSP) Systems

Digital processing of analog signals (mixed signal applications) forms one of the most important applications of DSP theory.

A/Dconverter

DSPsystem

D/Aconverter

...101011... …001010...

analog input

digital input

analog output

digital output

antialiasingprefilter

samplingand

quantization

discreteto

continuous

reconstructionfilter

Page 2: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 2

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Spectral Representation of Continuous-Time Signals

Fourier Analysis:

– f represents frequency in units of cycles/second or Hz and represents frequency in units of radians/second.

– If x(t) is a voltage signal, then X( and X(f) have units of volts/Hz.

– The conversion between frequency variables is = 2f.

Fourier Synthesis:

dtetxX tj

)()( dtetxfX ft2j

)()(or

2d

eXtx tj

)()( or dfefXtx ft2j

)()(

Page 3: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 3

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Some Important Fourier Pairs

Constant (DC) signal: (this signal contains only DC or zero

frequency)

Impulse: (this signal contains all frequencies)

Complex exponential (sinusoid): (this signal contains only one

frequency component - in fact, this signal is used to define frequency!)

Real sinusoid: (this signal contains two frequency components, +/- f0)

These pairs are for frequency measured in Hz. Remember the following rule for changing variables with impulses:

)( fAA

1t )(

)( 0tf2j ffe 0

)()()cos( 0j

0j

0 ffe21

ffe21

tf2

)()(

2f2

f

Page 4: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 4

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Signals are “Sums of Sinusoids”

Periodic signals contain only discrete frequency components that are multiples of the fundamental frequency.

Non-periodic signals contain a continuous set of frequency components.

The amplitude and phase of eachsinusoid forms the spectrum of the signal!

Page 5: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 5

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Linear, Time-Invariant (LTI) Systems

Impulse response:

Convolution:

Frequency Response:

t t0 0

)(t)(th

)(tx

dxthty )()()(

)cos( ft2 )(cos)( fHft2fH

)()( fHth

Page 6: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 6

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Filtering

H(f) modifies the amplitude of the input signal’s spectrum according to :

H(f) modifies the phase of the input signal’s spectrum according to:

)( fX )()()( fXfHfY

)()()( fXfHfY

)()()( fHfXfY

Page 7: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 7

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Review of Sampling

A bandlimited signal is one whose frequency spectrum contains no components greater than some maximum frequency fmax.

The sampling theorem states that bandlimited signals can be reconstructed perfectly from their samples provided the sampling rate fs (in samples/second) satisfies:

2 fmax is called the Nyquist rate.

0 fmax-fmax f f0

maxf2fs

X(f) X(f)

Page 8: EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing

08-Sept, 98 EE421, Lecture 1 8

EE421, Fall 1998Michigan Technological University

Timothy J. Schulz

Aliasing

Sampling at a rate slower than the Nyquist rate will result in aliasing. That is, frequency components greater than fs / 2 will be folded back into the Nyquist interval. This is generally a bad thing.

0 fs-fsf (Hz)

0 fs-fsf (Hz)

Don’t let this happen to you!