Digital Signal Processing: L1
Digital Signal Processing
Topic Marks
Minor 20
Attendance + Quiz 5
Lab Minor 10
Attendance 5
Major 15
Project 10
Major 35
Total 100
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Signals
•Flow of information
•Measured quantity that varies with time (or
position)
•Electrical signal received from a transducer
(microphone, thermometer, accelerometer,
antenna, etc.)
•Electrical signal that controls a process
•Continuous-time signals: voltage, current,
temperature, speed, . . .
•Discrete-time signals: daily
minimum/maximum temperature, lap intervals
in races, sampled continuous signals, . . .
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Signal Processing
Signals may have to be transformed in order to
Amplify or filter out embedded information
Detect patterns
Prepare the signal to survive a transmission channel
Prevent interference with other signals sharing a medium
Undo distortions contributed by a transmission channel
Compensate for sensor deficiencies
Find information encoded in a different domain
To do so, we also need
Methods to measure, characterize, model and simulate transmission channels
Mathematical tools that split common channels and transformations into easily
manipulated building blocks
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Analog Electronics in DSP
Passive networks (resistors, capacitors,
inductances, crystals, SAW filters),
non-linear elements (diodes, . . . ),
(roughly) linear operational amplifiers
Advantages:
• Passive networks are highly linear over a very large
dynamic range and large bandwidths
• Analog signal-processing circuits require little or no
power
• Analog circuits cause little additional interference
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Digital Signal Processing
Analog/digital and digital/analog converter, CPU, DSP, ASIC, FPGA.
Advantages:
Noise is easy to control after initial quantization
Highly linear (within limited dynamic range)
Complex algorithms fit into a single chip
Flexibility, parameters can easily be varied in software
Digital processing is insensitive to component tolerances, aging, environmental conditions,
electromagnetic interference
But:
Discrete-time processing artifacts (aliasing)
Can require significantly more power (battery, cooling)
Digital clock and switching cause interference
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Typical DSP applications
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Typical DSP applications
M1. Signals and systems: Discrete sequences and systems, their types and properties. Linear time-
invariant systems, convolution.
M2. Fourier transform: Phasors as orthogonal base functions. Forms of the Fourier transform,
convolution theorem, impulse combs in the time and frequency domain.
M3. Random sequences and noise: Random variables, stationary processes, autocorrelation,
crosscorrelation, deterministic crosscorrelation sequences, filtered random sequences, white noise,
exponential averaging.
M4. Discrete sequences and spectra: Periodic sampling of continuous signals, periodic signals,
aliasing, sampling and reconstruction of low-pass and band-pass signals, IQ representation of band-
pass signals, spectral inversion.
M5. Finite and infinite impulse-response filters: Properties of filters, implementation forms, window-
based FIR design, use of frequency-inversion to obtain highpass filters, use of modulation to obtain
band-pass filters, FFT-based convolution, polynomial representation, z-transform, zeros and poles, use
of analog IIR design techniques (Butterworth, Chebyshev I/II, elliptic filters).
M6. Discrete Fourier transform: Continuous versus discrete Fourier transform, symmetry, linearity,
review of the FFT, real-valued FFT.
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Digital Signal Processing: L2
Sequences and Systems
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Sequences and Systems
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Some Simple Sequences
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Properties of sequences
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Units and decibel
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Units and decibel
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Units and decibel
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Types of Discrete Systems
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Types of Discrete Systems
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Examples:
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Examples:
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Examples:
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Constant-coefficient difference equations
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Constant-coefficient difference equations
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Constant-coefficient difference equations
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Digital Signal Processing: L3
Fourier Transform for Periodic and Discrete Signals
The nature of the time-domain signal (discrete or continuous, and periodic or not), is reflected in
the nature of the spectrum.
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Digital Signal Processing
•Real world is always analog and continuous.
•Physical phenomena are time-limited and not truly periodic in a mathematical sense.
•Digital Signal Processing (which runs on digital computers) will by definition always work with
discrete-time signals and discrete-frequency signals.
•The mere act of analog to digital conversion (ADC) removes the continuous character of a time-
domain signal.
• DSP programs describe, by definition, a periodic process.
• DSP programs transform an infinite stream of samples, applying the same type of processing to
each sample.
•The spectral analysis tool implemented by a DSP program is a DFT.
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Discrete Fourier Transform
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Discrete Fourier Transform
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Duality Property
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Fourier Transform Pair
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Convolution and Multiplication
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Types of Discrete Systems
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Development of the Discrete Fourier Transform
•The sampling operation in the time domain is expressed a multiplication with a pulse train.
•Some aliasing may unavoidable in case the sample frequency is too low. 10/63
Development of the Discrete Fourier Transform
•We will bound the time-domain signal in
time, to a window of N samples.
•This is implemented by time-windowing
the domain-domain pulse stream.
•In the frequency domain, we will convolve
the periodic continuous spectrum with a
sinc spectrum corresponding to the time
window.
•The resulting effect is a ringing in the
spectrum. This ringing is caused by the
truncation of the time domain signal.
•The stronger the truncation, the heavier
the ringing.
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Development of the Discrete Fourier Transform
•Finally, time-domain representation
periodic, so that we get a discrete and
periodic time-domain waveform as well
as a periodic spectrum.
• This requires a time-domain
convolution with a pulse waveform
spectrum, and corresponds to sampling
of the spectrum in the frequency
domain.
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Constant-coefficient difference equations
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Constant-coefficient difference equations
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Constant-coefficient difference equations
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Digital Signal Processing: L4
Convolution
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Convolution
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Properties of Convolution
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Proof: all LTI systems just apply convolution
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Proof: all LTI systems just apply convolution
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Proof: all LTI systems just apply convolution
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Proof: all LTI systems just apply convolution
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Filter
A filter is a circuit capable of passing (or amplifying) certain frequencies while attenuating other frequencies.
Thus, a filter can extract important frequencies from signals that also contain undesirable or irrelevant
frequencies.
In the field of electronics, there are many practical applications for filters. Examples include:
Radio communications: Filters enable radio receivers to only "see" the desired signal while rejecting all
other signals (assuming that the other signals have different frequency content).
DC power supplies: Filters are used to eliminate undesired high frequencies (i.e., noise) that are present on
AC input lines. Additionally, filters are used on a power supply's output to reduce ripple.
Audio electronics: A crossover network is a network of filters used to channel low-frequency audio to
woofers, mid-range frequencies to midrange speakers, and high-frequency sounds to tweeters.
Analog-to-digital conversion: Filters are placed in front of an ADC input to minimize aliasing. 9/63
Types of Filters
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Passive and Active Filters
• Passive filters include only passive components- resistors, capacitors, and inductors.
Passive filters are most responsive to a frequency range from roughly 100 Hz to 300 MHz. The limitation on
the lower end results from the fact that the inductance or capacitance would have to be quite large at low
frequencies. The upper-frequency limit is due to the effect of parasitic capacitances and inductances. Careful
design practices can extend the use of passive circuits well into the gigahertz range.
• Active filters use active components, such as op-amps, in addition to resistors and capacitors, but not
inductors.
Active filters are capable of dealing with very low frequencies (approaching 0 Hz), and they can provide
voltage gain (passive filters cannot). Active filters can be used to design high-order filters without the use of
inductors; this is important because inductors are problematic in the context of integrated-circuit
manufacturing techniques. However, active filters are less suitable for very high-frequency applications
because of amplifier bandwidth limitations. Radio-frequency circuits must often utilize passive filters.
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Key Points and Terms
•Response curves are used to describe how a filter behaves. A response curve is simply a graph
showing an attenuation ratio (VOUT / VIN) versus frequency.
•Attenuation is commonly expressed in units of decibels (dB).
•Frequency can be expressed in two forms: either the angular form ω (units are rad/s) or the more
common form of f (units of Hz, i.e., cycles per second). These two forms are related by ω = 2πf.
•Finally, filter response curves may be plotted in linear-linear, log-linear, or log-log form.
•The most common approach is to have decibels on the y-axis and logarithmic frequency on the x-axis.
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Key Points and Terms
-3 dB frequency (f3dB). This term, pronounced "minus 3dB frequency", corresponds to the input
frequency that causes the output signal to drop by -3dB relative to the input signal.
•The -3 dB frequency is also referred to as the cutoff frequency.
•It is the frequency at which the output power is reduced by one-half (which is why this frequency is
also called the "half-power frequency"), or the output voltage is the input voltage multiplied by 1/√2.
•For low-pass and high-pass filters, there is only one -3 dB frequency. However, there are two -3 dB
frequencies for band-pass and notch filters—normally referred to as f1 and f2.
Center frequency (f0). The center frequency, a term used for band-pass and notch filters, is a central
frequency between the upper and lower cutoff frequencies. The center frequency is commonly defined as
the arithmetic mean (see equation below) or the geometric mean of the lower and upper cutoff frequency.
Bandwidth (β or B.W.). The bandwidth is the width of the passband, and the passband is the band of
frequencies that do not experience significant attenuation when moving from the input of the filter to the
output of the filter.
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Key Points and Terms
Stopband frequency (fs). This is a particular frequency at which the attenuation reaches a specified
value.
•For low-pass and high-pass filters, frequencies beyond the stopband frequency are referred to as
the stopband.
•For band-pass and notch filters, two stopband frequencies exist. The frequencies between these two
stopband frequencies are referred to as the stopband.
Quality factor (Q): The quality factor of a filter conveys its damping characteristics. In the time
domain, damping corresponds to the amount of oscillation in the system’s step response. In the
frequency domain, higher Q corresponds to more (positive or negative) peaking in the system’s
magnitude response. For a bandpass or notch (bandstop) filter, Q represents the ratio between the
center frequency and the -3dB bandwidth (i.e., the distance between f1 and f2).
For both band-pass and notch filters:
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Digital Signal Processing: L4
Convolution
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Convolution
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Properties of Convolution
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Proof: all LTI systems just apply convolution
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Proof: all LTI systems just apply convolution
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Proof: all LTI systems just apply convolution
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Proof: all LTI systems just apply convolution
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Filter
A filter is a circuit capable of passing (or amplifying) certain frequencies while attenuating other frequencies.
Thus, a filter can extract important frequencies from signals that also contain undesirable or irrelevant
frequencies.
In the field of electronics, there are many practical applications for filters. Examples include:
Radio communications: Filters enable radio receivers to only "see" the desired signal while rejecting all
other signals (assuming that the other signals have different frequency content).
DC power supplies: Filters are used to eliminate undesired high frequencies (i.e., noise) that are present on
AC input lines. Additionally, filters are used on a power supply's output to reduce ripple.
Audio electronics: A crossover network is a network of filters used to channel low-frequency audio to
woofers, mid-range frequencies to midrange speakers, and high-frequency sounds to tweeters.
Analog-to-digital conversion: Filters are placed in front of an ADC input to minimize aliasing. 9/63
Types of Filters
10/63
Passive and Active Filters
• Passive filters include only passive components- resistors, capacitors, and inductors.
Passive filters are most responsive to a frequency range from roughly 100 Hz to 300 MHz. The limitation on
the lower end results from the fact that the inductance or capacitance would have to be quite large at low
frequencies. The upper-frequency limit is due to the effect of parasitic capacitances and inductances. Careful
design practices can extend the use of passive circuits well into the gigahertz range.
• Active filters use active components, such as op-amps, in addition to resistors and capacitors, but not
inductors.
Active filters are capable of dealing with very low frequencies (approaching 0 Hz), and they can provide
voltage gain (passive filters cannot). Active filters can be used to design high-order filters without the use of
inductors; this is important because inductors are problematic in the context of integrated-circuit
manufacturing techniques. However, active filters are less suitable for very high-frequency applications
because of amplifier bandwidth limitations. Radio-frequency circuits must often utilize passive filters.
11/63
Key Points and Terms
•Response curves are used to describe how a filter behaves. A response curve is simply a graph
showing an attenuation ratio (VOUT / VIN) versus frequency.
•Attenuation is commonly expressed in units of decibels (dB).
•Frequency can be expressed in two forms: either the angular form ω (units are rad/s) or the more
common form of f (units of Hz, i.e., cycles per second). These two forms are related by ω = 2πf.
•Finally, filter response curves may be plotted in linear-linear, log-linear, or log-log form.
•The most common approach is to have decibels on the y-axis and logarithmic frequency on the x-axis.
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Key Points and Terms
-3 dB frequency (f3dB). This term, pronounced "minus 3dB frequency", corresponds to the input
frequency that causes the output signal to drop by -3dB relative to the input signal.
•The -3 dB frequency is also referred to as the cutoff frequency.
•It is the frequency at which the output power is reduced by one-half (which is why this frequency is
also called the "half-power frequency"), or the output voltage is the input voltage multiplied by 1/√2.
•For low-pass and high-pass filters, there is only one -3 dB frequency. However, there are two -3 dB
frequencies for band-pass and notch filters—normally referred to as f1 and f2.
Center frequency (f0). The center frequency, a term used for band-pass and notch filters, is a central
frequency between the upper and lower cutoff frequencies. The center frequency is commonly defined as
the arithmetic mean (see equation below) or the geometric mean of the lower and upper cutoff frequency.
Bandwidth (β or B.W.). The bandwidth is the width of the passband, and the passband is the band of
frequencies that do not experience significant attenuation when moving from the input of the filter to the
output of the filter.
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Key Points and Terms
Stopband frequency (fs). This is a particular frequency at which the attenuation reaches a specified
value.
•For low-pass and high-pass filters, frequencies beyond the stopband frequency are referred to as
the stopband.
•For band-pass and notch filters, two stopband frequencies exist. The frequencies between these two
stopband frequencies are referred to as the stopband.
Quality factor (Q): The quality factor of a filter conveys its damping characteristics. In the time
domain, damping corresponds to the amount of oscillation in the system’s step response. In the
frequency domain, higher Q corresponds to more (positive or negative) peaking in the system’s
magnitude response. For a bandpass or notch (bandstop) filter, Q represents the ratio between the
center frequency and the -3dB bandwidth (i.e., the distance between f1 and f2).
For both band-pass and notch filters:
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Digital Signal Processing: L6
Time Domain Representation of LTI Systems
• If the input-output relationship of a system is represented in time domain or as
a function of time.
• Impulse response of an LTI system is defined as the output of the system due
to unit impulse input applied at time t=0 or n=0.
2
Discrete Convolution
Let,
3
Discrete Convolution
Let,
4
Discrete Convolution: Graphical Method
5
Discrete Convolution: Graphical Method
6
Discrete Convolution: Graphical Method
7
Discrete Convolution: Graphical Method
8
Discrete Convolution: Graphical Method
9
Discrete Convolution: Graphical Method
10
Discrete Convolution: Graphical Method
11
Properties of Convolution
Commutative Property
Proof:
12
Properties of Convolution
Associative Property
13
Properties of Convolution
Distributive Property
14
Properties of Convolution
Multi-linearity Property
15
Properties of Convolution
Multi-linearity Property
16
Properties of Convolution
Time-Shift Property
17
Properties of Convolution
Differentiation Property
18
Properties of Convolution
Impulse Convolution Property
19
Properties of Convolution
20
Digital Signal Processing: L7
Review: DTFT
2
Review: DTFT
3
DFT
4
DFT
5
DFT
6
DFT-Example
7
DFT-Example
8
Inverse Discrete Fourier Transform
9
Inverse Discrete Fourier Transform
10
Inverse Discrete Fourier Transform
11
DFT
12
DFT
13
DFT
14
DFT: Periodicity
15
DFT: Periodicity
16
DFT: Circular Time Shift
17
DFT: Circular Time Shift
18
DFT: Conjugation
19
DFT: Conjugation
20
DFT: Circular Frequency Shift
21
DFT: Circular Frequency Shift
22
DFT: Multiplication
23
DFT: Multiplication
24
DFT: Multiplication
25
Digital Signal Processing: L9
Circular Convolution
2
Circular Convolution
Concentric Circle Method
3
Circular Convolution: Concentric Circle Method
4
Circular Convolution: Concentric Circle Method
5
Circular Convolution: Concentric Circle Method
6
Circular Convolution: Concentric Circle Method
7
Circular Convolution: Concentric Circle Method
8
Circular Convolution: Matrix Method
9
Circular Convolution: Matrix Method
10
Circular Convolution: Matrix Method
11
Circular Convolution: Table Method
12
Digital Signal Processing: L10
Overlap-Add Method
2
Overlap-Add Method
3
Overlap-Add Method
4
Overlap-Add Method
5
Overlap-Add Method
6
Overlap-Add Method
7
Overlap-Add Method
8
Overlap-Add Method
9
Overlap-Save Method
10
Overlap-Save Method
11
Overlap-Save Method
12
Overlap-Save Method
13
Overlap-Save Method
14
Overlap-Save Method
15
Overlap-Save Method
16
Overlap-Save Method
17
Overlap-Save Method
18
Overlap-Save Method
19
Digital Signal Processing: L11
Overlap-Add Method
2
Overlap-Add Method
3
Overlap-Add Method
4
Overlap-Add Method
5
Overlap-Add Method
6
Overlap-Add Method
7
Overlap-Add Method
8
Overlap-Add Method
9
Overlap-Save Method
10
Overlap-Save Method
11
Overlap-Save Method
12
Overlap-Save Method
13
Overlap-Save Method
14
Overlap-Save Method
15
Overlap-Save Method
16
Overlap-Save Method
17
Overlap-Save Method
18
Overlap-Save Method
19
Overlap-Save Method
20Findthe output y(n) of a filter whose impulse response is h(n) ={1,1, 1} and input signal
x(n)={3,-1,0,1,3,2,0,1,2,1} using
(i)overlap-save method
(ii) overlap-add method.
20
Overlap-Save Method
21
Overlap-Save Method
22
Overlap-Add Method
23
Overlap-Add Method
24
Overlap-Add Method
2. Using linear convolution find y(n)=x(n) * h(n) for the sequence x(n)={1,2,-1,2,3,-2,-3,-1,1,1,2,-1} and h(n)={1,2}
Compare the result by solving the problem using (a)overlap-save method (b)overlap-add method.
25
Digital Signal Processing: L12
Correlation of Discrete-Time Signals
•A signal operation similar to signal convolution, but with completely different physical meaning, is signal
correlation.
•The signal correlation operation can be performed either with one signal (autocorrelation) or between two
different signals (crosscorrelation).
•Physically, signal autocorrelation indicates how the signal energy (power) is distributed within the signal,
and as such is used to measure the signal power.
•Typical applications of signal autocorrelation are in radar, sonar, satellite,and wireless communications
systems.
2
Correlation of Discrete-Time Signals
•Devices that measure signal power using signal correlation are known as signal correlators.
•There are also many applications of signal crosscorrelation in signal processing systems, especially when
the signal is corrupted by another undesirable signal (noise) so that the signal estimation (detection) from a
noisy signal has to be performed.
•Signal crosscorrelation can be also considered as a measure of similarity of two signals.
3
Correlation of Discrete-Time Signals
4
Correlation of Discrete-Time Signals
5
Correlation of Discrete-Time Signals
6
Correlation of Discrete-Time Signals
7
Correlation of Discrete-Time Signals
8
Correlation of Discrete-Time Signals
9
Digital Signal Processing: L12
The Discrete Fourier Transform (DFT)
2
The Discrete Fourier Transform (DFT)
3
The Discrete Fourier Transform (DFT)
4
The Discrete Fourier Transform (DFT): Multiplications
5
The Discrete Fourier Transform (DFT): Additions
6
Total Operation Count
7
Operation Count Makes DFT Impractical
8
The Divide and Conquer Approach
9
The Divide and Conquer Approach
10
The Decimation in Time (DIT) Algorithm
11
The Decimation in Time (DIT) Algorithm
12
The Decimation in Time (DIT) Algorithm
13
Butterfly Diagram
14
The Decimation in Time (DIT) Algorithm
15
The Decimation in Time (DIT) Algorithm
16
The Decimation in Time (DIT) Algorithm
17
“Divide and Conquer” Results in Savings!
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“Divide and Conquer” Results in Savings!
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“Divide and Conquer” Results in Savings!
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“Divide and Conquer” Results in Savings!
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“Divide and Conquer” Results in Savings!
22