Chap1 2

1 Introduction
Signals, Systems, and Digital Signal Processing

Definition Basic Elements of a Digital Signal Processing Advantages of Digital over Analog Signal Processing
Backgrounds1
Classification of Signals
Multi-channel and Multi-dimensional Signals Continuous-Time versus Discrete-Time Signals Continuous-Valued versus Discrete-Valued Signals
Concepts of Frequency
Physical Interpretation of Signal Frequency Continuous-Time Sinusoidal Signals Discrete-Time Sinusoidal Signals
NCTU/CSIE/DSP LAB Audio Processing Group
1.1 Signals, Systems, & Digital Signal Processing
Backgrounds2
Definition Basic Elements of DSP Advantages of Digital over Analog Signal Processing
Definition
Signals
Any physical quantity that varies with time, space or any other independent variable. Communication beween humans and machines.
Backgrounds3
Systems
mathematically a transformation or an operator that maps an input signal into an output signal. can be either hardware or software. such operations are usually referred as signal processing.
Digital Signal Processing

The representation of signals by sequences of numbers or symbols and the processing of these sequences.
Basic Elements of a Digital Signal Processing System

A/D Converter
Converts an analog signal into a sequence of digits Analog Input Signal
A/D A/D Converter Converter
t 0 {3, 5, 4, 6 ...}
Backgrounds4
Digital Input Signal
D/A Converter
Converts a sequence of digits into an analog signal
Digital Digital Signal Signal Processing Processing
D/A D/A Converter Converter
Analog Output Signal
Digital Output Signal
Advantages of Digital over Analog Processing

Better control of accuracy Easily stored on magnetic media Allow for more sophisticated signal processing Cheaper in some cases
Backgrounds5
1.2 Classification of Signals

Multichannel versus Multidimensional Signals
Backgrounds6
Signals may be generated by multiple sources or multiple sensors. Such signals are multi-channel signals. A signal which is a function of M independent variables is called multidimensional signals.
Continuous-Time versus Discrete-Time Signals

Continuous-time signals are defined for every value of time. Discrete -time signals are defined at discrete values of time.
Continuous-Valued versus Discrete-Valued Signals

A signal which takes on all possible values on a finite range or infinite range is said to be a multi-channel signal. A signal takes on values from a finite set of possible values is said to be a multi-dimensionall signal.
Examples
A picture is a two-dimensional signal
I(x,y) is a function of two variables.
Backgrounds7
A black-and-white television picture is a three-dimensional signal

I(x,y,t) is a function of three variables.
A color TV picture is a three-channel, three-dimensional signals

Ir(x,y,t), Ig(x,y,t), and Ib(x,y,t)
1.3 Concepts of Frequency

Backgrounds8
Physical Interpretation of Signal Frequency

A familiar term in physics and mathematics
Radio transmitter/receiver Amplifier Color photography ...............
Backgrounds9

Interpretation
Closed related to a specific type of periodic motion called harmonic oscillation, described by sinusoidal functions. Usually a dimension of inverse time.
Backgrounds10
Why is the term important ?

Time

Time Domain Representation
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An Observation
Fourier Transform
Frequency-Domain Representation
Backgrounds12
Signal can be represented either through time- or frequency-domain. Frequency-domain representation of signals provides another viewpoint benefitial to signal analysis, human sensitivity, system design, and phenomenon interpretation. Frequency Transform: the tool to decompose a timedomain signal into frequency components. The "frequency" can be considered as the varying rate of the signal f(x) in x-domain.
f(t)
f(F) Time
Spectrum Frequency
1.3 Concepts of Frequency

Backgrounds13
Continuous-Time Sinusoidal Signals

Backgrounds14
Definition Xa(t) = A cos( t+ )
-*<t<*
A is the amplitude of the sinusoid
is the frequency in radians per second
is the phase in radians
F=/2 is the frequency in cycles per second or hertz
Time
Continuous-Time Sinusoidal Signals (Cont.)

For every fixed value of F, Xa(t) is periodic
Xa(t+Tp) = Xa(t), Tp=1/F
Backgrounds15
Continuous-time sinusoidal signals with distinct frequencies are themselves distinct Increasing the frequency F results in an increase in the rate of oscillation
Discrete-Time Sinusoidal Signals

Definition X(n) = A cos( n+ ), -*<t<*
A is the amplitude of the sinusoid is the frequency in radians per second is the phase in radians
Backgrounds16
f=/2 is the frequency in cycles per second or hertz

X(n) = A cos( n+ )
Discrete-Time Sinusoidal Signals(Cont.)

A discrete-time sinusoidal is periodic only if its frequency f is a rational number
X(n+N) = X(n), N=p/f, where p is an integer
Backgrounds17
X(n) = A cos( n+ )
Discrete-time sinusoidal signals where frequencies are separated by an integer multiple of 2 are identical
X1(n) = A cos( 0 n) X2(n) = A cos( (0 +2) n)
The highest rate of oscillation in a discrete-time sinusoidal is attained when = or (=-), or equivalently f=1/2.
X(n) = A cos(( 0+)n) = -A cos((0+)n
2. The Process of A/D and D/A Conversion

Basic Elements Signal Sampling Anti-aliasing Filtering Quantization Interpolator Smoothing Filters
Backgrounds18
2.1 Basic Elements

Analog Input Signal
Cut-off Frequency Fc
Antialiasing Antialiasing Filter Filter
t 0 t t
Backgrounds19
Sampling Frequency Fs
Sampler Sampler
The number of bits

Quantizer Quantizer
{3, 5, 4, 6 ...}
{3, 5, 4, 6 ...}
Smoothing Smoothing Filter Filter
Interpolator Interpolator
Cut-off Frequency Fc*
Sampling Frequency Fs*
2.1 Basic Elements(c.1)

An Observation
Backgrounds20
The mapping between discrete-frequency and analogfrequency is one-to many


Sampler
Converts a continuous-time signal into a discrete-time signal.
Backgrounds21
Anti-aliasing Filter(A low-pass filter)

Deletes the frequency components above a threshold frequency to avoid the aliasing effects.
Quantizer
Converts a discrete-time continuous-valued signal into a discrete-time, discrete-valued signal

t 0
Sampler Sampler
Quantizer Quantizer {3, 5, 4, 6 ...}

t
Fc
Fs
t
2.1 Basic Elements

Interpolator
Converts a discrete-time signal into a continuous-time signal.
Backgrounds22
Smoothing Filter
Deletes the frequency components above a threshold frequency to avoid the image signal.
t t
{3, 5, 4, 6 ...}
t

Analog Input Signal
t 0 t t
Backgrounds23
Sampler Sampler
The number of bits

Quantizer Quantizer & &Coder Coder
{3, 5, 4, 6 ...}
t t
{3, 5, 4, 6 ...}
2.2 Signal Sampling

Sampling
Backgrounds24
The conversion of a continuous-time signal into a discrete-time signal obtained by taking "samples" of the continuous-time signal at discrete-time instants Xa(nT) = X(n) where T is the sampling interval
Many-to-One Mapping between F and f
Time X(t) X(n)
Analog Frequency <==> Discrete FrequencyBackgrounds25

The relationship between the time variables t and n
t = nT = n/Fs
2.2 Signal Sampling (c.1)
Analog Frequency F (or ) <==> Discrete Frequency f ()

Xa(nT) = x(n) = Acos(2FnT +) = A cos (2pnF/Fs + ) compare with x(n) = A cos (2fn+) f = F/Fs or = T
f = F/Fs or = T
f
X(n) = A cos( n+ )

Frequency Restriction
Backgrounds26
Continuous-Time Frequency -<F< -*<< Discrete-Time Frequency - 1/2 < f < 1/2 -<< Relation and Restriction - Fs/2 < F < Fs/2 - Fs < < Fs
Many-to-One Mappling between F and f

Frequency Relation
Backgrounds27
Many-to-one Mapping Fk = F0 + kFs are indistinguishable from the frequency F0 after resampling and hence they are aliased of F0.
f 0 -Fs/2 -Fs Fs/2 Fs F
Folding Frequency ==> Fs/2

Sampling Theorem
If the highest frequency contained in an analog signal Xa(t) is Fmax =B and the signal is sampled at a rate Fs > 2Fmax = B, then Xa(t) can be exactly recovered from its sample values using the interpolation
Backgrounds28
sin 2 Bt g(t ) = 2 Bt
Thus Xa(t) may be expressed as
n Xa ( n / Fs ) g( t ) x a (t ) = n Fs =
where Xa(n/Fs) = Xa(nT) = X(n) are the sample of Xa(t)

History
Backgrounds29
Cauchy, French, 1841 Functions could be nonuniformly sampled and averaged over a long period. Whittaker, Scottish, 1915 A bandlimited function can be completely reconstructed from samples. (first mathematical proof of a general sampling theorem) K. Ogura, Japanese, 1920 If a function is sampled at a frequency at least twice the highest function frequency, the samples contain all the information in the function, and can reconstruct the function. Carson, American, 1920 Unpublished proof that related the same result to communication.

History (c.1)
Backgrounds30
Nyquist, Sweden, 1928 For complete signal reconstruction, the required frequency bandwidth is proportional to the signalling speed. The minimum bandwidth is equal to half the number of code elements per second. Expressed the theorem in terms that are familiar to communication engineers. Kotelnikov, Russian, 1933 A proof of sampling theorem Shannon, American, 1949 Unified many aspects of sampling and founded the larger science of information theory.
2.3 Antialiasing Filters

Aliasing
F +- kFs are mapped into the same discrete frequency f 0 Fs/2 F Fs
Backgrounds31
-Fs -Fs/2
2.3 Antialiasing Filters

1 Low-Pass Filter
Backgrounds32
Purpose: Delete the frequency components that will be aliased to low frequency components.
Fc
Low-Pass Filters Fc < Fs/2
2.4 Quantization
z Quantization
Express each sample value as a finite number of digits.
Backgrounds33
Output of Sampler
z Quantization Error
The error introduced in representing the continuous-value signal by a discrete value levels. Output of Quantization
z Signal-to-quantization noise ratio, SQNR(dB)

1.76 + 6.02b 16 bits CD audio data has a quality of more than 96 dB
2.4 Quantization (c.1)

SQNR(dB)
The maximum root mean square signal Srms is
Backgrounds34
S rms =
Q 2 b 1
21 / 2
The rms quantization error is

2 E rms = e p e de ( )
1/ 2
1 = Q
e 2de
1/ 2
Q 2 = 12
1/ 2
Q
( 12 )1 / 2
The poweer ratio is

S 3 ( dB ) = 10 log ( 2 2b ) = 6 . 02b + 1. 76 E 2

Observation
The quantization error is random and perceptually similar to analog white noise for large amplitude signals. Problems low-amplitude signals. narrow band signals.
Backgrounds35
Dither
Decorrelates the errors from the signals. Allows the digital system to encode amplitude smaller than the LSB.
2.4 Quantization-- Types of Dither

Gaussian pdf
Contributes to Q2/4
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Rectangular pdf
Contributes to Q2/12
Triangular pdf
Contributes to Q2/6

Earliest Dither in Word War II
Jim MacArthur has pointed out Bombers used mechanical computers to perform navigation and bomb trajectory calculations. These computers perform more accurately when flying on board the aircraft and less well on ground. Engineers realized that the vibration from the aircraft reduced the error from sticky moving parts.
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Optimal Interpolator
2.5 Interpolator
Optimal Interpolator:
from Sampling Theorems
Zero-order Interpolator
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First-order Interpolator
no distortion for the frequency components below Fs/2 no frequency components above Fs/2 exist and smoothing filtering is not necessary
n Xa (n / Fs)g(t ) xa (t ) = n Fs =
Suboptimal Interpolator
Signal Mangitude Spectrum F Fs 2Fs
distortion exists for the frequency components below Fs/2 result in passing frequencies above the folding frequency and smoothing filtering is necessary
2.6 Smoothing Filters
Backgrounds39
Delete the frequency components above a threshold frequency to avoid the image signal introduced by suboptimal filters
Low-pass filtering 1 Low-Pass Filter Smoothing Smoothing Filter Filter
t t
Fc'
Signal Mangitude Spectrum

F
2Fs
Fs
2Fs
2.7 Concluding Remarks

Time/Frequency Illustrarion
Antialiasing filtering and Antiimaging filtering
Backgrounds40
2.7 Concluding Remarks

Analog Input Signal
t 0 t t
Backgrounds41
Sampler Sampler
The number of bits

Quantizer Quantizer
{3, 5, 4, 6 ...}
{3, 5, 4, 6 ...}

Experiment 1
WinAmp Architectures
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Describe the functionality of Input Plugin, Output Plugin, DSP Plugin, and VIS Plugin.
Find the Input Plugin for Wave File. Change the decoded results for Stereo Channels as
L[n] = L[n] + {L[n] R[n]} R[n] = R[n] + {R[n] L[n]}

Find the suitable parameters for the two parameters. Describe the noise you have found during the experiments.
Experiment 2
Sampling Rates Change Problem.
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1. Change the sampling rates from 44.1kHz to 22.05 kHz by eliminate the odd samples of stereo channels. L[n] = L[2n] R[n] = R[2n] Listen to the resulted music and describe the artifacts. Compare the spectrum through COOL editor to find the spectrum artifact. 2. Change again the sampling rates from 44.1 kHz to 11.025 kHz by three samples every four samples. L[n] = L[4n] R[n] = R[4n] Listen to the resulted music and describe the artifacts. Compare the spectrum through COOL editor to find the spectrum artifact.
Experiments
Analog Input Signal
Cut-off Frequency Fc Antialiasing Antialiasing Filte r Filter
t 0
Backgrounds44
Sampler Sampler
The number of bits

Quantizer Quantizer
{3, 5, 4, 6 ...}
Cut-off Frequency Fc* Smoothing Smoothing Filter Filter
t
Fc=Fc' and is below 1.5 k ==> Lowpass filtering effects Fc > Fs/2 ==> Aliasing effects Quantization effects Fs' > Fs or Fs' < Fs ==> Frequency mismatching Fc' > Fs/2 ==> Image effects
Hearing Area in Frequency Domain

Backgrounds45
1. Blind Deconvolution for the first music 2. Piano Music a. Original One b. Low-pass One c. Image Distortation (too many high frequency) d. Aliasing effects Quantization Noise is independent of the Original Signals ?

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Chap1 2

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1 Introduction

Signals, Systems, and Digital Signal Processing

NCTU/CSIE/DSP LAB Audio Processing Group

1.1 Signals, Systems, & Digital Signal Processing

NCTU/CSIE/DSP LAB Audio Processing Group

Digital Signal Processing

Basic Elements of a Digital Signal Processing System

Digital Input Signal

Digital Digital Signal Signal Processing Processing

D/A D/A Converter Converter

Analog Output Signal

Digital Output Signal

NCTU/CSIE/DSP LAB Audio Processing Group

Advantages of Digital over Analog Processing

NCTU/CSIE/DSP LAB Audio Processing Group

1.2 Classification of Signals

Continuous-Time versus Discrete-Time Signals

Continuous-Valued versus Discrete-Valued Signals

NCTU/CSIE/DSP LAB Audio Processing Group

A black-and-white television picture is a three-dimensional signal

A color TV picture is a three-channel, three-dimensional signals

NCTU/CSIE/DSP LAB Audio Processing Group

1.3 Concepts of Frequency

NCTU/CSIE/DSP LAB Audio Processing Group

Physical Interpretation of Signal Frequency

NCTU/CSIE/DSP LAB Audio Processing Group

Physical Interpretation of Signal Frequency

Why is the term important ?

NCTU/CSIE/DSP LAB Audio Processing Group

Physical Interpretation of Signal Frequency

Physical Interpretation of Signal Frequency

NCTU/CSIE/DSP LAB Audio Processing Group

1.3 Concepts of Frequency

NCTU/CSIE/DSP LAB Audio Processing Group

Continuous-Time Sinusoidal Signals

Definition Xa(t) = A cos( t+ )

is the phase in radians

F=/2 is the frequency in cycles per second or hertz

NCTU/CSIE/DSP LAB Audio Processing Group

Continuous-Time Sinusoidal Signals (Cont.)

NCTU/CSIE/DSP LAB Audio Processing Group

Discrete-Time Sinusoidal Signals

f=/2 is the frequency in cycles per second or hertz

NCTU/CSIE/DSP LAB Audio Processing Group

Discrete-Time Sinusoidal Signals(Cont.)

NCTU/CSIE/DSP LAB Audio Processing Group

2. The Process of A/D and D/A Conversion

NCTU/CSIE/DSP LAB Audio Processing Group

2.1 Basic Elements

The number of bits

Digital Digital Signal Signal Processing Processing

Smoothing Smoothing Filter Filter

Analog Output Signal

Cut-off Frequency Fc*

Sampling Frequency Fs*

NCTU/CSIE/DSP LAB Audio Processing Group

2.1 Basic Elements(c.1)

The mapping between discrete-frequency and analogfrequency is one-to many

2.1 Basic Elements(c.2)

Anti-aliasing Filter(A low-pass filter)

Antialiasing Antialiasing Filter Filter

Quantizer Quantizer {3, 5, 4, 6 ...}

2.1 Basic Elements