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Brief Notes On 1 Part of ME579: - Would Like To See X (F), The Fourier Transform of X (T), - C - We Have

The document discusses discrete Fourier transforms (DFT) and digital signal processing concepts. It notes that DFT can be used to calculate Fourier series coefficients or the Fourier transform of a sampled signal. The DFT yields discrete frequency values that are periodic in both time and frequency. Windowing, sampling, and zero-padding are also covered as they relate to the DFT and transforming between time and frequency domains.

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Will Black
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72 views

Brief Notes On 1 Part of ME579: - Would Like To See X (F), The Fourier Transform of X (T), - C - We Have

The document discusses discrete Fourier transforms (DFT) and digital signal processing concepts. It notes that DFT can be used to calculate Fourier series coefficients or the Fourier transform of a sampled signal. The DFT yields discrete frequency values that are periodic in both time and frequency. Windowing, sampling, and zero-padding are also covered as they relate to the DFT and transforming between time and frequency domains.

Uploaded by

Will Black
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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Brief Notes on 1st Part of ME579

Would like to see X(f), the Fourier Transform of x(t), or ck, the Fourier Series coefficients, if the time history is periodic. We have: x(n ) samples of x(t) every seconds for n=0,1,2,3.N-1. And can do Discrete Fourier Transforms (DFT) to give: Xk for k=0,1,2,.N-1 corresponding to frequencies fk = 0, fs/N, 2fs/N, .(N-1)fs/N.

Brief Notes on 1st Part of ME579


ISSUES: We have done three things WINDOWED
(TRUNCATION - LEAKAGE)

xw(t) = w(t) . x(t) Xw(f) = W(f) convolved with X(f) SAMPLED IN TIME

(ALIASING, CLIPPING, QUANTIZATION NOISE)

Periodic in Frequency.

xS(t) = (t-n) . x(t)

Brief Notes on 1st Part of ME579


SAMPLED IN FREQUENCY

Xk = XS(f) evaluated at f= fk

fk = 0, fs/N, 2fs/N, .(N-1)fs/N.

Signal now becomes Periodic in Time

xn+qN = xn

Domains: Time and Frequency


Sampled/Discrete Multiplication Real and Even Real and Odd Narrow Length (T or fs) Periodic Convolution Real and Even Imaginary and Odd Broad Resolution (1/T or )

When do we Zero Pad Signals?


1. When result of an operation should yield a longer signal than original signal(s). e.g. convolution and time-delay. 2. When we want to have a clearer picture of Xs(f), the Fourier Transform of the sampled signal, xs(t). [Would prefer to transform more data to get better resolution, i.e., a spikier W(f). Use zero padding when we dont have any more data to transform.]

Discrete Fourier Transform (DFT), Xk Relationship to X(f) and ck


Assume no aliasing when sampling [fs > 2 f max]

Have x(n) for n=0,1,2,N-1; Xk = D.F.T.(x); k=0,1,2,3,N-1.

Periodic Signals: N = a whole number of periods = q Tp ck = Xk/N for -(N/2) < k < (N/2) Transients (some aliasing will occur): X(f) Xk for -fs/2 < f < fs/2

Discrete Fourier Transform (DFT), Xk


Symmetric about 0 and fs/2.
Real Part: even symmetry about these points Imaginary Part: odd symmetry about these points

Periodic

Xk for k=+(N/2),+(N/2)+1, . N-1 equal to Xk for k=-(N/2),-(N/2)+1, . 1 [fftshift will rearrange for you; you have to make the corresponding frequency vector.]

Analog to Digital - Digital to Analog


Input/Output Range, No. of Bits, fs Analog to Digital Conversion (ADC) Quantization Error, Clipping, Sample and hold, Aliasing, Anti-aliasing filters, Sample rate. f1 = highest frequency of interest fc = filter cut-off frequency fmax = highest frequency in filtered signal fs > 2 fmax = sample rate.

f1 < fc < fmax < fs/2

Analog to Digital - Digital to Analog


Digital to Analog Conversion Similar issues as for ADC Zero-order hold characteristics sinc function in frequency, distorts signal in range -fmax<f<fmax, distortion less as fs >>> fmax. Reconstruction Filter: fmax < fc << fs/2.

Other Things
We use the delta function, (t) or (f), in a lot of our theory and proofs.
Sifting property in integrals Integral from - to + of exp(j2ft) Sampling theory

We looked at the FFT algorithm (not on exam). Convolution of continuous and discrete signals.

All the Transforms: timefrequency


Complex Fourier Series x(t) ck periodic in time, discrete in frequency Fourier Transforms x(t) X(f) continuous in time and frequency Fourier Transform of a sampled signal: xs(t) Xs(f) OR x(n) Xs(f) discrete in time, periodic in frequency Discrete Fourier Transform (finite set of data used) xn Xk; n and k: 0,1,2..N-1. periodic and discrete in both time and frequency Xs(f) = (1/ )

X(f - q fs)

Using the Discrete Fourier Transform


Fourier Series coefficients
(no aliasing and N corresponds to a whole no. of periods) ck = Xk/N for k=0,1,2N/2.

Approximate the Fourier Transform X(f) of x(t) for frequencies: f = k.fs/N


(no aliasing)

X(f)

at f = k.fs/N

= Xk

for k=0,1,2(N/2).

Sampled version of X s(f) Xs(f)

(sampled signal, xs(t) was of finite length = N points)


at f = k.fs/N

= Xk

Can zero pad to evaluate at more frequency points

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