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Interpolation

Yao Wang

Polytechnic University, Brooklyn, NY11201

http://eeweb.poly.edu/~yao

Outline

– A/D and D/A converters

• Sampling

– Nyquist sampling theorem

– Aliasing due to undersampling:

• temporal and frequency domain interpretation

• Sampling sinusoid signals

• Reconstruction from samples

– Reconstruction using sample-and-hold and linear interpolation

– Frequency domain interpretation (sinc pulse as interpolation kernel)

• Sampling rate conversion

– Down sampling

– Up sampling

– Demonstration

Analog to Digital Conversion

1

T=0.1

Q=0.25

0.5

-0.5

-1

A2D_plot.m

Two Processes in A/D Conversion

Sampling Quanti-

zation

xc(t) x[n] = xc(nT) x[ n]

Sampling Quantization

Period Interval

T Q

• Sampling: take samples at time nT

– T: sampling period;

x[ n] = x( nT ),−∞ < n < ∞

– fs = 1/T: sampling frequency

– Q: quantization interval or stepsize

xˆ[ n] = Q[ x(nT )]

How to determine T and Q?

– A fast varying signal should be sampled more frequently!

– Theoretically governed by the Nyquist sampling theorem

• fs > 2 fm (fm is the maximum signal frequency)

• For speech: fs >= 8 KHz; For music: fs >= 44 KHz;

• Q depends on the dynamic range of the signal amplitude and

perceptual sensitivity

– Q and the signal range D determine bits/sample R

• 2R=D/Q

• For speech: R = 8 bits; For music: R =16 bits;

• One can trade off T (or fs) and Q (or R)

– lower R -> higher fs; higher R -> lower fs

• We consider sampling in this lecture, quantization in the next

lecture

Nyquist Sampling Theorem

• Theorem:

– If xc(t) is bandlimited, with maximum frequency fm (or ωm =2π fm )

– and if fs =1/T > 2 fm or ωs =2π /T >2 ωm

– Then xc(t) can be reconstructed perfectly from x[n]= xc(nT) by

using an ideal low-pass filter, with cut-off frequency at fs/2

– fs0 = 2 fm is called the Nyquist Sampling Rate

• Physical interpretation:

– Must have at least two samples within each cycle!

Temporal Domain Interpretation:

Sampling of Sinusoid Signals

Sampling above

Nyquist rate

ωs=3ωm>ωs0

Reconstructed

=original

Sampling under

Nyquist rate

ωs=1.5ωm<ωs0

Reconstructed

\= original

Aliasing: The reconstructed sinusoid has a lower frequency than the original!

Frequency Domain

Interpretation of Sampling

Original signal

Sampling

impulse train The spectrum of the

sampled signal includes

the original spectrum and

its aliases (copies) shifted

Sampled signal to k fs , k=+/- 1,2,3,…

ωs>2 ωm The reconstructed signal

from samples has the

frequency components

upto fs /2.

Sampled signal

ωs<2 ωm When fs< 2fm , aliasing

(Aliasing effect) occur.

Sampling of Sinusoid in Frequency

Domain

Spectrum of

cos(2πf0t)

-f0 0 f0

No aliasing

fs >2f0

fs -f0 >f0

Reconstructed

-fs -f0 -fs -fs+f0 -f0 0 f0 fs-f0 fs fs+f0

signal: f0

-fs/2 fs/2

With aliasing

f0<fs <2f0 (folding) 0

fs -f0 <f0

Reconstructed signal: fs -f0 -fs -f0 -fs -f0 -fs+f0 fs-f0 f0 fs fs+f0

With aliasing

fs <f0 (aliasing) -fs

0 fs

f0-fs <f0

Reconstructed signal: fs -f0 -fs -f0 -f0 -f0+fs f0-fs f0 fs+f0

More examples with Sinusoids

demo

– Aliasing: fs < fm (perceived frequency: fm -fs )

– Folding: fm < fs < 2fm (perceived frequency: fs -fm )

– No need to distinguish these two phenomena. Both lead to a

false frequency lower than the original frequency

Strobe Movie

How to determine the necessary sampling

frequency from a signal waveform?

should be at least two samples in the shortest ripple

• The inverse of its length is approximately the highest

frequency of the signal

Fmax=1/Tmin

Tmin

samples in this

interval, in order not

to miss the rise and

fall pattern.

Sampling with Pre-Filtering

Pre-Filter Periodic

H (f) Sampling

x(t) x’(t) xd(n)

Sampling

period T

• To prevent aliasing, pre-filter the continuous signal so that fm<fs/2

• Ideal filter is a low-pass filter with cutoff frequency at fs/2

(corresponding to sync functions in time)

•Common practical pre-filter: averaging within one sampling interval

How to Recover Continuous Signals

from Samples?

– Sampling and hold (rectangular kernels)

– Linear interpolation (triangular kernels)

– High order kernels

– Ideal kernel: sinc function

Sample-and-Hold vs. Linear

Interpolation

1.2

0.6

Reconstructed signal by linear interpolation

0.4

0.2

0

0 1 2 3 4 5 6 7 8 9 10

Reconstruction Using Different

Kernels

“reconstruction”

Frequency domain interpretation

Original signal

Sampled signal

ωs>2 ωm

Ideal

reconstruction filter

(low-pass)

Reconstructed signal

(=Original signal)

Ideal Interpolation Filter

Hr(f)

T

f

- fs/2 fs/2

T | f |< f s / 2 sin π t / T

H r ( f )= hr (t ) =

0 otherwise π t /T

∞ ∞

sin[ π (t − nT ) / T ]

xr (t ) = xs (t ) ∗ hr (t ) = ∑ x[n] hr (t − nT ) = ∑ x[n]

π (t − nT ) / T

n =−∞ n =−∞

Sampling Rate Conversion

sampling rate

– Necessary for image display when original image size differs

from the display size

– Necessary for converting speech/audio/image/video from

one format to another

– Sometimes we reduce sample rate to reduce the data rate

• Down-sampling: reduce the sampling rate

• Up-Sampling: increase the sampling rate

Down-Sampling Illustration

1.2

0.8

Down-sampling by a

0.6

factor of 2 = take

0.4

every other sample

0.2

0 To avoid aliasing of

0 1 2 3 4 5 6 7 8 9 10

any high frequency

content in the

original signal,

1.2

should smooth the

1

original signal before

0.8 down-sampling --

0.6 Prefiltering

0.4

0.2

0

0 1 2 3 4 5 6 7 8 9 10

Down Sampling by a Factor of M

– T’=MT, fs’=fs/M

• Should apply a prefilter to limit the bandwidth of the original signal to

1/M-th of the original

• Without prefiltering, aliasing occur in the down-sampled signal.

• Ideal prefilter: low pass filter with cut-off frequency at 1/M (maximum

digital frequency=1, corresponding to fs/2)

• Practical filter: averaging or weighted average over a neighborhood

Low-pass filter

Gain: M M

Cut-off: 1/M

x [n] x’[n] xd [n] = x [nM]

Down-Sampling Example

– Original sequence: 1,3,4,7,8,9,13,15…

– Without prefiltering, take every other sample:

• 1,4,8,13,…

– With 2-sample averaging filter

• Filtered value=0.5*self+0. 5*right, filter h[n]=[0.5,0.5]

• Resulting sequence:

– 2, 5.5,8.5,14,…

– With 3-sample weighted averaging filter

• Filtered value=0.5*self+0.25*left+0.25*right, filter h[n]=[0.25,0.5,0.25]

• Resulting sequence (assuming zeros for samples left of first):

– 1.25, 4.5,8,12.5,…

Upsampling by linear interpolation

1.2

0.8

0.6

0.4

0.2

0

0 1 2 3 4 5 6 7 8 9 10

1.2

0.8

0.6

0.4

0.2

0

0 1 2 3 4 5 6 7 8 9 10

Missing samples need to be filled from neighboring available samples using interpolation filter

Up Sampling by a Factor of L

• T’=T/L, fs’=fs*L

– The estimation of the missing samples from original samples is

known as interpolation

• Interpolation can be decomposed into two steps

– Zero-padding: insert L-1 zeros in between every two samples

– Low-pass filtering: to estimate missing samples from neighbors

– Simplest interpolation filter: linear interpolation

Lowpass Filter

L Gain = L

Cutoff = 1 / L

x[n] xe [n] xi [n]

period T padding period T’ = T/L period T’ = T/L

Up-Sample Example

of 2

– Original sequence: 1,3,4,7,8,9,13,15…

– Zero-padding:

• 1,0,3,0,4,0,7,0,…

– Sample and hold

• Repeat the left neighbor, filter h[n]=[1,1]

• 1,1,3,3,4,4,7,7,…

– With linear interpolation

• New sample=0.5*left+0. 5*right, filter h[n]=[0.5,1,0.5]

• Resulting sequence:

– 1,2,3,3.5,4,5.5,7,8,…,…

Demonstration

different pre-filters, and up-sampling with different

interpolation filters

• Compare both sound quality and frequency spectrum

• Matlab code (sampling_demo.m)

Down-2 followed by up-2, both using good filters

y psd-y

0.4 0

0.2

-20

0

-0.2 -40

-0.4

-60

5000 10000 15000 0 5000 10000

0.4 0 0

0.2 -20

-20 0.4

-40

0

0.2

-40 -60

-0.2

0 -80

-0.4

-60

2000 4000 6000 0 5000 10000 -10 0 10 0 0.2 0.4 0.6 0.8

0.4 0

0

1

0.2

-20

0 0.5 -50

-0.2 -40

0

-0.4 -100

-60

5000 10000 15000 0 5000 10000 -5 0 5 0 0.2 0.4 0.6 0.8

Down-2 followed by up-2, both using bad filters

y psd-y

0.4 0

0.2

-20

0

-0.2 -40

-0.4

-60

5000 10000 15000 0 5000 10000

0 0

0.2 -10

-20 0.4

-20

0

0.2 -30

-40

-0.2

-40

0

-0.4 -60 -50

2000 4000 6000 0 5000 10000 0 0.2 0.4 0.6 0.8

0

0

1

0.2

-20 -20

0 0.5

-40

-40

-0.2

0 -60

-0.4 -60

5000 10000 15000 0 5000 10000 -2-1 0 1 2 0 0.2 0.4 0.6 0.8

Down-4 followed by up-4, both using good filters

y psd-y

0.4 0

0.2

-20

0

-0.2 -40

-0.4

-60

5000 10000 15000 0 5000 10000

0 0

0.2

-20 0.4

0 -50

0.2

-40

-0.2

0

-60 -100

1000 2000 3000 0 5000 10000 -10 0 10 0 0.2 0.4 0.6 0.8

0

0

1

0.2

-20

-50

0 0.5

-40 -100

-0.2

0

-60 -150

5000 10000 15000 0 5000 10000 -10 0 10 0 0.2 0.4 0.6 0.8

Down-4 followed by up-4, both using bad filters

y psd-y

0.4 0

0.2

-20

0

-0.2 -40

-0.4

-60

5000 10000 15000 0 5000 10000

0 0

-10

0.2 0.4

-20 -20

0 -30

0.2

-40

-0.2 -40

0

-50

-60

1000 2000 3000 0 5000 10000 -1012 0 0.2 0.4 0.6 0.8

0

1 0

0.2

-20

-20

0 0.5

-40 -40

-0.2

0 -60

-60

5000 10000 15000 0 5000 10000 -4-3-2-101234 0 0.2 0.4 0.6 0.8

MATLAB Code

What Should You Know (I)

• Sampling:

– Know the minimally required sampling rate:

• fs > 2 fmax , Ts < T0,min /2

• Can estimate T0,min from signal waveform

signal is under-sampled.

– Can plot the spectrum of a sampled signal

• The sampled signal spectrum contains the original spectrum and its

replicas (aliases) at kfs, k=+/- 1,2,….

• Can determine whether the sampled signal suffers from aliasing

– Understand why do we need a prefilter when sampling a signal

• To avoid alising

• Ideally, the filter should be a lowpass filter with cutoff frequency at fs /2.

– Can show the aliasing phenomenon when sampling a sinusoid

signal using both temporal and frequency domain interpretation

What Should You Know (II)

• Interpolation:

– Can illustrate sample-and-hold and linear interpolation from

samples.

– Understand why the ideal interpolation filter is a lowpass filter with

cutoff frequency at fs /2.

– Know the ideal interpolation kernel is the sinc function.

– Interpolation using the sinc kernel is NOT required

– Know the meaning of down-sampling and upsampling

– Understand the need for prefiltering before down-sampling

• To avoid aliasing

• Know how to apply simple averaging filter for downsampling

– Can illustrate up-sampling by sample-and-hold and linear

interpolation

References

– Has good conceptual / graphical interpretation

(copies provided, Sec. 4.3,4.5 not required)

and sampling rate conversion. Sec. 1,2. (copy provided)

– Optional reading (More depth in frequency domain interpretation)

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