Coding Techniques for Analog Source

EGR 544 Communication Theory 4. Coding Techniques for Analog Signal Z. Aliyazicioglu Electrical and Computer Engineering Department Cal Poly Pomona ...
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EGR 544 Communication Theory

4. Coding Techniques for Analog Signal

Z. Aliyazicioglu Electrical and Computer Engineering Department Cal Poly Pomona

Coding Techniques for Analog Source • There are several analog source coding techniques • Most of the coding techniques are applied speech and image coding Three type of analog source encoding • Temporal Waveform coding :design to represent digitally the time-domain characteristic of the signal • Spectral waveform coding: signal waveform is sub divided into different frequency band and either the time waveform in each band or its spectral characteristics are encoded. • Model-based coding: Based on the mathematical model of source.

Cal Poly Pomona

Electrical & Computer Engineering Dept.

EGR 544-2 2

1

Temporal Waveform Coding Most common used methods: • Pulse-code modulation (PCM) • Differential pulse-code modulation (DPCM) • Delta modulation(DM) Pulse-code modulation (PCM) Let’s have continuous source function x (t ) and each sample taken from x (t ) is xn at sampling rate fs ≥ 2W, where W is the highest frequency in x (t ) . In PCM, each sample is quantized to one of 2R amplitude level, where number of binary digits used to represent each sample. The bit rate will be Rfs [bit/s] Cal Poly Pomona

Electrical & Computer Engineering Dept.

EGR 544-2 3

Pulse-code modulation (PCM) • The quantized value will be xn and xn = xn − qn

qn quantization error

• Assume that a uniform quantizer is used, then PDF of quantization error is 1  p( q) =  ∆  0

if −

∆ ∆ ≤q≤ 2 2 o.w.

∆ is step size and obtained ∆ = 2− R

Cal Poly Pomona

Electrical & Computer Engineering Dept.

EGR 544-2 4

2

Pulse-code modulation (PCM) Mean square value of the quantization error (or noise) power is ∆/2

E ( q2 ) = ∫

−∆ / 2

q 2 p( q) dq =

1 2 1 −2 R 2 ∆ = 12 12

Mean square value of the quantization error power in dB

E ( q 2 ) dB = 10log

1 2 1 ∆ = 10log 2 −2 R = −10.8 − 6 R [dB] 12 12

Quantization noise decreases by 6dB/bit Quantization noise for 8 bit -58.8 dB It can be measured by signal-to-quantization noise ratio (SQNR) in dB SQNR = 10log

P (average power of source signal) [dB] E(q 2 )

If source is sinusoidal Cal Poly Pomona

SQNR = −1.76 + 6.02 R [dB]

Electrical & Computer Engineering Dept.

EGR 544-2 5

Pulse-code modulation (PCM) • The non-uniform quantizer characteristic can be obtained by passing the signal through a non-linear device the compress the signal amplitude • For example: µ-law compressor: A Logarithmic compressor input-output function

y =

log(1 + µ x ) log(1 + µ )

µ is a parameter that gives desired compression µ=225 selected for USA and Canada. µ=225 , it will drop quantization noise power about –77dB for 7 bit quantization Cal Poly Pomona

Electrical & Computer Engineering Dept.

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3

Example of µ -law

µ=20

µ=100 Cal Poly Pomona

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Differential pulse-code modulation (DPCM) • The differences between samples are expected to be smaller than the actual sampled amplitude value. • The simple solution is to encode the differences between successive samples rather than the samples themselves. • Fever bits require to represent the differences

 Let xn denote the current sample from the source and let xn denote the predicted value of xn, defined as p

xˆn = ∑ ai xn −1 i =1

 • xn is weighted linear combination of the past p samples and {ai} are the predicted coefficient that are selected to minimize the  error between xn and xn

Cal Poly Pomona

Electrical & Computer Engineering Dept.

EGR 544-2 8

4

Differential pulse-code modulation (DPCM)  • The mean square error between xn and xn is given 2 p    ε p = E (e ) = E  xn − ∑ ai xn−i   i =1    2 n

p

p

i =1

i =1

= E ( xn2 ) − 2∑ ai E ( xn xn −i ) + ∑

p

∑ a a E( x i

j =1

x

n −i n − j

j

)

• Selecting {ai } to minimize the MSE Assume that source output is stationary and φ(m) shows the autocorrelation function of xn p

p

i =1

i =1

ε p = φ (0) − 2∑ aiφ (i ) + ∑ • To minimize εp set

j =1

i

j

p

∑ a φ (i − j ) = φ ( j ) i =1

Cal Poly Pomona

p

∑ a a φ (i − j )

i

Electrical & Computer Engineering Dept.

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Differential pulse-code modulation (DPCM) • If autocorrelation function is not known, it may be estimated as

1 φˆ(n) = N

p

N −n

∑xx i =1

xˆn = ∑ ai xn −i

Predicted output is

i i+n

difference

en = xn − xˆn

i =1

p

en = xn − ∑ ai xn −i i =1

en − en = en − ( xn − xˆn ) = e + xˆ − x

Predicted value of xn

n

xˆn = ∑ ai xn −i i =1

Cal Poly Pomona

n

n

= xn − xn = qn

p

Encoder Electrical & Computer Engineering Dept.

Quantization error

EGR 544-2 10

5

Differential pulse-code modulation (DPCM) xn = xn + qn

The quantized sample xn differs from the input xn by quantization error qn

To low-pass filter

Decoder

Cal Poly Pomona

Electrical & Computer Engineering Dept.

EGR 544-2 11

Delta Modulation en = ( xn − xˆn ) To transmitter

Predicted (estimated) value

qn = en − en

xˆn = xn−1 = xˆn −1 + en −1 xˆn = xn−1 + qn−1

= en − ( xn − xˆn ) Source Encoder Output

Source Decoder Cal Poly Pomona

Electrical & Computer Engineering Dept.

EGR 544-2 12

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Delta Modulation Equivalent realization of Delta modulation To transmitter

Source Encoder

Output

Source Decoder Cal Poly Pomona

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Delta Modulation • The performance of the DM encoder is limited by two types of distortion • Slope overload distortion • Step size is too small • Granular noise • Step size is to large Granular noise

Slope-overload distortion

Cal Poly Pomona

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Delta Modulation •

Alternative solution is variable step size: Step size is increased when the waveform has steep slope and decreased when the waveform has a relatively small slope

• One of the method is called continuous variable slope delta modulation (CVSD) If en , en−1 ,and en−2 has same sign

∆ n = α∆ n −1 + k1 Granular noise

Otherwise

∆ n = α∆ n −1 = k2 where

0 0

k =1

The difference between xn and xˆn

The filter coefficients {ak } can be selected to minimize the mean square error

p

en = xn − xˆn = xn − ∑ ak xn −k k =1

Cal Poly Pomona

Electrical & Computer Engineering Dept.

EGR 544-2 21

Encoding methods for Speech signal • Speech signal band limits 200-3200Hz. • Sampling frequency 8000samples/s for all encoder except DM Encoding method

Quantization

Coder

Transmission rate(bits/s)

PCM Log PCM DPCM ADPCM DM ADM

Linear Logarithmic Logarithmic Adaptive Binary Adaptive Binary

12 bits 7-8 bits 4-6 bits 3-4 bits 1 bit 1 bit

96,000 36,000-64,000 32,000-48,000 24,000-32,000 32,000-64,000 16,000-32,000

LPC/CELP

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2400-9600

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