Sampling Signals

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sampling – creating a discrete signal from a continuous process. ¯ downsampling (decimation) .... Frequency Domain An
Sampling Signals Overview:

 We use the Fourier transform to understand the discrete sampling 

and re-sampling of signals. One key question is when does sampling or re-sampling provide an adequate representation of the original signal?

Terminology:

 sampling – creating a discrete signal from a continuous process.  downsampling (decimation) – subsampling a discrete signal  upsampling – introducing zeros between samples to create a longer signal

 aliasing – when sampling or downsampling, two signals have same sampled representation but differ between sample locations. Matlab Tutorials: samplingTutorial.m, upSample.m

320: Sampling Signals

c A.D. Jepson and D.J. Fleet, 2005

Page: 1

Sampling Signals I(x)

continuous signal x

I[n]





sampled signal

● ●



● ●



● ●













● ●

n g[m]

downsampling





● ●



I[n]









2

g[m]

m f[n]

upsampling





● ●







● ●



320: Sampling Signals





g[m]

● ●









2

f[n]

n

Page: 2

Up-Sampling Consider up-sampling a signal I [n] of length N :

 Increase number of samples N by a factor of ns.  Step 1. Place ns 1 zeros after every sample of I [n], to form f [n] 0

of length nsN , namely

(

f0 [n] =

I [n=ns ]

for n mod ns

0

otherwise.

= 0;

(1)

Raw Upsampled Signal

Signal

1.5

2

1

1.5 1

0.5 0

0

f

I[n]

0.5

−0.5

0

−0.5

−1

−1

−1.5 −2 0

5

10 n

320: Sampling Signals

15

−1.5 0

5

10

15 n

20

25

30

Page: 3

Up-Sampling (Cont.)

 Step 2. Interpolate signal f : f = S  f ; for some smoothing kernel S [n]: 0

0

(2)

Here, in order for f [n] to interpolate f0[n], we require that S [0] = 1 and S [jns ] = 0 for nonzero integers j . Sinc Filtered Upsampled Signal (b) 1.5 1

f[n]

0.5 0 −0.5 −1 −1.5 0

5

10

15 n

20

25

30

 Many smoothing kernels can be used (see upSample.m).

320: Sampling Signals

Page: 4

Frequency Analysis of Up-Sampling Step 1. Fourier transform of the raw up-sampled signal f0[n]:

F (f [n])[k] 

X

Nns

0

=

n=0 N 1

X

1

X

N

i!ks n

f0 [n]e

=

1

s

f0 [jns ]ei!k jns

j =0 s

I [j ]ei!k jns

j =0

Here !ks

2

= Nn k , s

F (f )[k] =

X

N

1

0

so the last line above becomes 2

I [j ]ei N kj

j =0

 F (I )[k];

N ns

for

2

 k < N2ns :

(3)

Since F (I ) is N -periodic, equation (3) implies that F (f0) consists of ns copies of

F (I ) concatenated together.

Amplitude Spectrum of Raw Upsampled Signal 6

5

5

4

4 |FT|

|FT|

Amplitude Spectrum of I[n] 6

3

3

2

2

1

1

0

−5

0 k

320: Sampling Signals

5

0 −15

−10

−5

0 k

5

10

15

Page: 5

Fourier Analysis of Up-Sampling Step 2 Recall Step 2 is to form f [n]

S

=

 f , for some interpolation filter S . 0

However, notice from the inverse Fourier transform that (for N even) I [j ] =

=

1

X

N=2

1

2

I^[k ]ei N kj

N

k= N=2 N=2 ns

X

N ns

1

2 kjn i Nn s s

f^0 [k ]e

=

f [jns ]:

(4)

k= N=2

Here we used f (jns) = I (j ) in the last line. Notice the left term in the last line above is ns times the inverse Fourier transform of B [k ]f^0[k ] where B is the box function, B [k ]

= 1

N=2

for

k

< N=2 and

B [k ] = 0 otherwise. We can therefore evaluate this inverse Fourier

transform at every pixel n, and not just at the interpolation values jns, to construct a possible interpolating function f [n] f [n] = ns

F

1

(B (k )f^0 [k ]): Sinc Filtered Upsampled Signal (b)

Amp. Spec. (Upsampled Sig.(r), Sinc Filter(g), Filtered Sig.(b) 12

1.5 1

10

0.5 f[n]

|FT|

8 6

0

4

−0.5

2

−1

0 −15

(5)

−10

−5

320: Sampling Signals

0 k

5

10

15

−1.5 0

5

10

15 n

20

25

30

Page: 6

Down-Sampling Consider down-sampling a signal I [n] of length N :

 Reduce number of samples N by a factor of ns, where ns is a divisor of N .

 Define the comb function:

X

(N=ns )

C (n; ns) =

1

Æn;mns

m=0 I [n]

ns ●































ns = 4



n

 Step 1. Introduce zeros in I [n] at unwanted samples. g0 [n] = C [n; ns ]I [n]:

(6)

 Step 2. Downsample signal g : 0

g [m] = g0 [mns ]; for 0

320: Sampling Signals

 m < N=ns:

(7)

Page: 7

Frequency Domain Analysis of Down-Sampling Proposition 1. The Fourier transform of the comb function is another comb function:

F (C [n; ns]) = nN C [k; N=ns]:

(8)

s



C(ω) −π

2π ns

ns = 4 π ω

0

Recall the frequency ! and wave number k are related by ! the spacing in the plot above is !s

2

N

2

2

= N k.

So

= Nn = n . s s

Proposition 2. Pointwise product and convolution of Fourier transforms. Suppose f [n] and g [n] are two signals of length N (extended to be N -periodic). Then

F (f [n]g[n])

=



F (f )  F (g) N= 1 X ^

1

N

2

N

1

f [j ] g^[k

j ]:

(9)

j = N=2

where f^ and g^ denote the Fourier transforms of f and g , respectively. The proofs of these two propositions are straight forward applications of the definition of the Fourier transform given in the preceeding notes, and are left as exercises. 320: Sampling Signals

Page: 8

Fourier Analysis of Down-Sampling Step 1 Recall Step 1 is to form g0[n] = C [n; ns]I [n]. By Prop. 1 and 2 above, we have

F (g ) 0

= =

 =

F (C [n; ns]I [n]) N 1 C [; N=ns ]  I^[] ns N N= X 1 2

ns 1

ns

C [j ; N=ns ]I^[k

j = N=2+1 ns r0

X

I^[k

r= r0 +1

r

N ns

j]

];

(10)

where, due to the periodicity of I^[k ], we can use any integer r0 (eg. r0 = ns =2 for even ns ). ∧



I(ω)

−π

2π ns

g0(ω) 0

π ω

−π

0

ns = 4 π ω

Therefore g^0[k ] consists of the sum of replicas I^[k

rN=ns ] of the

Fourier transform of the original signal I , spaced by wavenumber N=ns or, equivalently, by frequency !s

2

= n . s

Note the Fourier transform g^0 has period 2 , so the contribution sketched above for !

  can be shifted 2 to the left.

320: Sampling Signals

Page: 9

Fourier Analysis of Down-Sampling Step 2 In Step 2 we simply drop the samples from g0[n] which were set to zero by the comb function C [n; ns]. That is g [m] = g0 [mns ]; for 0

 m < N=ns:

In terms of the Fourier transform, it is easy to show

F (g)[k] = g^[k] = g^ [k]; 0

N=(2ns )

for

Rewriting this in terms of the frequency !s;k

 k < N=(2ns): =

2 (N=ns )

k (note g [m] is

a signal of length N=ns ), and the corresponding frequency !k =

(11)

2

= Nk

!s;k ns of the longer signal g0 [n], we have g^[!s;k ] = g^0 [!k ] = g^0 [!s;k =ns ]; for ∧

f(ω)

ns = 4

−π

ωs

0



 !s;k < :

(12)

original signal

π ω



g(ω)

downsampled signal

−π

0

π

4π ω



f(ω) −π

320: Sampling Signals

upsampled and filtered signal 0

π ω

Page: 10

Nyquist Sampling Theorem Sampling Theorem: Let f [n] be a band-limited signal such that f^[! ] = 0

for all j! j > !0

for some !0 . Then f [n] is uniquely determined by its samples g [m]

=

f [m ns ] when !s =2 =

where 0

 ns

= 2=!0 .

> !0

or equivalently ns
0 f^[k ] = 0

for all j!k j > !0:

Then f (x) is uniquely determined by its samples I [m] = f (m  ) when