504209

Embedded
Signal
Processors

.
 Module I

Introduction to Real-Time Embedded Signal

Processing, Time-Domain Digital Signals,
Introduction to Digital Systems, Moving-Average
Filters: Structures and Equations, Digital Filters
Realization of FIR Filters, Nonlinear Filters
Implementation.
 Module II

Frequency-Domain Analysis and Processing, Discrete

Fourier Transform, Fast Fourier Transform, Simple
Low pass Filters Design and applications of Notch
Filters, Design of FIR Filters Design of IIR Filters,
Structures and Characteristics of IIR Filters,
Algorithms of Adaptive Filters, Design and Applications
of Adaptive Filters.
 Module III

Introduction to Digital signal processing systems, MAC, Barrel

shifter, ALU, Multipliers, Dividers, DSP processor architecture,
Software developments, Selections of DSP processors,
Hardware interfacing, DSP processor architectures: TMS
320C54XX, TMS 320C67XX Blackfin
processor: Architecture overview, memory management, I/O
management, On chip resources, programming considerations,
Real time mplementation Considerations, Memory System and
Data Transfer, Code Optimization
 Module IV

Practical DSP Applications: Audio Coding and Audio

Effects, Digital Image Processing, Two-Dimensional
Filtering, Image Enhancement, DTMF generation and
detection, FFT algorithms, Wavelet algorithms,
Adaptive algorithms: system identification, inverse
modeling, noise cancellation, prediction..
 Embedded Signal
Processors

• Laboratory Assignments/Experiments:
1. Design and simulate N point FFT by targeting suitable
DSP processor platform.
2.Design and simulate N tap FIR filter by targeting suitable
DSP processor platform.
3. Design and simulate LMS adaptive filter.
4. Performance comparison of different filter structures.
 Course
Outcomes:

1. The student will be capable of designing the system for linear filtering
using DFT.
2. The student will show skills for design of FIR and IIR filters for any
application.
3. The student will exhibit the knowledge of implementing DSP
algorithms on DSP
Processor Platforms.
4. The student will demonstrate the design of adaptive filters.
5. The student will demonstrate the ability to analyze filter structures.
EMBEDDED PROCESSOR
 characteristics in typical
embedded systems

1. Dedicated functions: usually executes a specific task

repeatedly.
2. Tight constraints: such as cost, processing speed, size,
and power consumption.
3. Reactive and real-time performance: must continuously
react to changes of the system’s input.
 selection of a suitable embedded
processor

• use fixed -function and hardwired processors. ASICs with DSP

capabilities but hardwired processors are very expensive to design
and produce.
• programmable processors, write software for the specific
applications. flexibility of writing different algorithms for different
products also fast upgrading the code .
• programmable digital signal processor- include many advanced
features higher clock speed, multiple multipliers and arithmetic
units, coprocessors for control and communication tasks, and
advanced memory configuration.
 programmable embedded signal
processors

• micro signal architecture (MSA).

• meet the computational demands and power constraints of
today’s embedded audio, video, and communication
• both DSP and microcontroller functionalities
• ability to execute highly complex DSP algorithms and basic
control tasks
• Efficient and dynamic power management
 Cont.

• two multiply-add units, two arithmetic-logic units, and a single

shifter.
• reduce-instruction-set-computer (RISC) for both control and
signal processing applications.
• set of multifunction instructions
• special instructions support video and wireless applications.
• fixed-point processor.
 REAL-TIME EMBEDDED
SIGNAL PROCESSING

• special design considerations

• two types of real-time system: hard and soft
• challenges when designing a hard real-time system.
– 1. Understanding DSP concepts and algorithms.
– 2. Resource availability.
– 3. Arithmetic precision.
– 4. Response time requirements.
– 5. Integration of software and hardware in embedded system.
 Time-Domain Signals
and Systems

• examine real-world DSP applications

• each experiment requires a set of general problem-solving
skills and related DSP principles,
• TIME-DOMAIN DIGITAL SIGNALS
• digital signal x(n) is defined as a function of time index n
time at nTs seconds sampled from an analog signal x(t)
with the sampling period Ts seconds.
 Sinusoidal Signals

• analog sine wave

• X(t)= A sin(2π ft ) = a sin (Ωt) ,
• sampling rate fs,

must satisfy the Nyquist sampling

frequency components higher than fs/2 will fold back to the frequency range
from 0 Hz to fs/2 Hz, which results in a distortion called aliasing.
 Random Signals

• sine wave is a deterministic signal defined exactly by a

mathematical equation
• speech, music, and noise are random signals. often corrupted by
noises
• different techniques based on the characteristics of signals and
noise to reduce the undesired noise
• useful measures associated with a random signal are mean,
variance, and autocorrelation functions.
• mean (or expected value) is independent of
time and is used to defined variance
 DIGITAL SYSTEMS

• processing of digital signals is combinations of fundamental

operations including addition (or subtraction), multiplication, and
time shift (or delay). ( have adders, multipliers and delay unit)
• digital filter alters the spectral content of input signals in a specified
manner.
• Common filtering objectives include removing noises, improving
signal quality, extracting information from signals, and separating
signal components. A digital filter is a mathematical algorithm that
can be implemented in digital hardware and software and operates
on a digital input signal
• A digital filter can be classified as being linear or nonlinear, time
invariant or time varying.
 Moving-Average Filters:
Structures and Equations

• An L-point moving-average filter is defined as

• requires L − 1 additions and L memory locations for storing signal
sequence
 Digital Filters

• bl are the filter coefficients.

• Defined a unit impulse function as

• output is called the impulse response of the filter, h(n),

• length of the impulse response is L; type of filter is called a finite

impulse response (FIR) filter.
• better performance can be achieved by using different filter
Coefficients derived from filter design techniques.
 Realization of FIR Filters

• basic building blocks.

• FIR filtering defined linear convolution of two sequences, h(n) and
x(n),
• Therefore, there are four types of symmetric FIR filters depending
on whether L is even or odd and whether coefficients bl have
positive or negative symmetry.
• symmetric coeffiients

• antisymmetric (or negative symmetric),

• multiplications required to implement the symmetric FIR filtering

reduced to half if L is even. only have to store half the amount (L/2)
of coefficients
 NONLINEAR FILTERS

• simple nonlinear filter, the median filter, which is very effective for
reducing impulse
• for reducing impulse noise is the use of a nonlinear median filter.
• An L-point running median filter is a nonlinear algorithm that does
not obey the superposition property
• Implemented by sliding window of odd length 2k+1
• Y[n] = med { x[n-k] ,……., x[n-1],x[n],x[n+1],……,x[n+k]}
 Frequency-Domain Analysis
and Processing

• z-transform, system concepts, and discrete Fourier transform with

their applications.
• simple time-domain techniques such as moving-average filters,
Hanning filters, and nonlinear median filters for removing noises
that corrupted the desired signals.
• used a moving-average filter with length L = 5, 10, and 20
• to enhance a sine wave corrupted by white noise. Found that this
filter worked for L = 5 and 10, but failed for L = 20. We also found
that the filter caused undesired effects such as attenuation of the
desired sine wave amplitude and shifting of the phase of signal.
• Not able to remove those narrowband noises.
• failed to attenuate a tonal noise in desired broadband signals such
as speech.
 z-TRANSFORM

• where z is a complex variable. The set of z values for which X(z)

exists is called the region of convergence. We can recover x(n) from
X(z) by using the inverse z- transform. For a causal sequence (i.e.,
x(n) = 0 for n < 0), the summation starts from n = 0. In addition, for a
finite-length causal sequence x(n), n = 0, 1, . ,N − 1, the summation
starts from n = 0 and ends at n = N − 1
 System Concepts

• If y(n) is the result of linear convolution of two sequences x(n) and

h(n) as expressed the z-transform of y(n) is given as
• Y(z)= X(z)H(z)= H(z)X(z)
• the convolution in the time domain is equivalent to the multiplication
in the z-domain.
• The transfer (or system) function of a given system expressed as
H(z) = Y(z)/X(z).
 A system that is both linear and time invariant is called a linear time-
invariant (LTI) system. This system can be represented in both the time
domain and the z-domain
 Digital Filters

• two classes of digital filters : FIR and IIR filters.

• In practical application, an FIR filter is implemented by using the
direct-form structure (or realization)
• Signal buffer {x(n), x(n − 1), x(n − 2), . . . , x(n − L + 1)} is also
called a delay buffer or a tapped-delay line,
• finite length of the impulse response guarantees that the FIR filters
are stable.
• In addition, a perfect linear-phase response can be easily designed
• disadvantage of FIR filters is the computational complexity,
because it may require a higher order filter to fulfill a given
frequency specification.
 IIR filter.

• If the impulse response of a filter is not a finite-length sequence, the

filter is called an IIR filter.
• transfer function as

• Coefficient sets {bl} and {am} are constants that determine the filter’s
• characteristics.
• direct-form I realization of the IIR filter can be requires two signal
buffers, treated as two FIR filters. combined into one by using the
direct-form II realization
• simple direct-form implementation of an IIR filter will not be used in
practical applications.

• To reduce this effect, a high-order IIR filter transfer function is factored

• into second-order sections plus a first -order section if the order of filter
is an odd number.

• These sections are connected in cascade or parallel to form an overall

• filter
 FREQUENCY ANALYSIS

• Frequency Response
• discrete-time Fourier transform (DTFT) of infinite-length digital
signal x(n) is

• evaluating the z-transform on the unit circle, |z|= 1, in the complex

z-plane is equivalent to the frequency-domain representation
• frequency-domain representation of the system as
 Discrete Fourier Transform

• If the digital signal x(n) is a finite-duration of length N, the DTFT

given in Equation modified to become the DFT

• where k is the frequency index. Usually the signal x(n) is a real-

valued sequence, but the DFT coefficients X(k) are complex values.
• DFT defined assumes that the signal is a periodic signal with
• period N. The DFT is equivalent to evaluating (or sampling) the
DTFT X(ω) at N equally spaced frequencies ωk = 2πk/N, k = 0, 1,
. . . , N − 1;
 limitation of DFT

• A limitation of DFT is its inability to handle signals extending over

all time. It is also unsuitable for analyzing nonstationary signals
(such as speech) that have time-varying spectra. For such a signal,
it makes more sense to divide the signal into blocks over which it
can be assumed to be stationary and estimate the spectrum of
each block.
 Fast Fourier
Transform

• Need- DFT that N complex multiplications and additions are

needed to produce one output. To compute N outputs, a total of
approximately N2 complex multiplications and additions are
required. A 1,024-point DFT requires over a million complex
multiplications and additions.
• family of very Efficient algorithms for computing the DFT.
• not a new transform that is different from the DFT; it is simply an effi
cient algorithm for computing the DFT by taking advantage of the
fact that many computations are repeated in the DFT because of
the periodic nature of the twiddle factors.
• The ratio of computing cost in terms of number of multiplications

• Each variant FFT has a different strength and makes different

trade-offs between code complexity, memory requirements, and
computational speed.
• The FFT algorithm becomes lengthy when N is not a power of 2.
This restriction on N can be overcome by appending zeros at the
tail of the sequence to cause N to become a power of 2
 Window
Functions

• a window is often employed for spectral analysis using DFT. For a

long signal sequence, taking only N samples for analysis is
equivalent to applying a rectangular window w(n) of length N to the
signal. This action can be expressed as

• Windowing not only produces leakage effects, it also reduces

• spectral resolution
 Simple Low-Pass Filters

• Assume that the signal x(n) consists of a sine wave corrupted by

random noise. Our goal is to develop and implement a digital low-
pass filter to reduce noise, thus enhancing the sinusoidal component.
• sinusoidal component (a peak) is located at a low frequency (200 Hz)
and can be enhanced by reducing noise components in high-
frequency ranges. This objective may be achieved by using a simple
low-pass filter.
• simple filters will work for the purpose of reducing random noise that
corrupts a low-frequency sinusoidal signal with high SNR.
 Design and Applications of Notch Filters

• A notch filter contains one or more deep notches in its magnitude

response. To create a notch at frequency ω0, we simply introduce a
pair of complex-conjugate zeros on the unit circle at angle ω0 as

• This is the FIR filter of order 2 because there are two zeros in the
system. a pair of complex-conjugate zeros guarantees that the filter
will have real-valued coefficients.
• The magnitude response of the second-order FIR notch filter has a
relatively wide bandwidth, which means that other frequency com-
• -ponents around the null are severely attenuated.
• To reduce the bandwidth of the null, introduce poles into the system.
Suppose that we place a pair of complex-conjugate poles at

• transfer function for the resulting filter is

 Design and Applications of Peak Filters

• moving-average FIR filter and the first-order IIR filter for enhancing
a sine wave that was corrupted by white noise. these filters have
undesired attenuation of signals because the gain is less than 1.
They also only provide about 10–15 dB of noise reduction,
• notch filters that have very narrow bandwidth for attenuation of
sinusoidal interference.
• Peak Filters enhance a sinusoidal signal that is corrupted by broad-
band noises.
• To create a peak (or narrow passband) at frequency ω0, by
introducing a pair of complex-conjugate poles on the unit circle at
angle ω0. However, the resulting second-order IIR filter will be
unstable. To solve this problem, we have to move the poles slightly
inside the unit circle (i.e., use rp < 1) as
• magnitude response of the second-order IIR peak filter has a relatively
wide bandwidth, which means that other frequency components around
the peak will also be amplified.
• To reduce the bandwidth of the peak, introduce zeros into the system.
Suppose that we place a pair of complex-conjugate zeros at radius rz
with the same angle as the poles; the transfer function for the resulting
second-order IIR filter is
• It is important that for designing a peak filter, the poles must be
closer to the unit circle as rp >rz
• The IIR filter defined can be applied as a simple parametric
• equalizer for boosting (rp > rz) or cutting (rp < rz) an audio signal.
The amount of boost or cut is determined by the difference between
rp and rz.
• The bandwidth of peaks or notches is determined by the value of rz.
 Digital Filtering

• INTRODUCTION - digital filters can be divided into two categories:

FIR and IIR. These filters can be represented by difference (or I/O)
equations, system transfer functions, and signal flow diagrams.

• The process of deriving the digital filter transfer function H(z) that
satisfies a given specification is called digital filter design.

• some applications require only simple filters such as the moving-

average, notch, and peaking filters.

• Linear, time-invariant filters are characterized by magnitude

response, phase response, stability, rise time, settling time, and
overshoot.
 Ideal Filters

• Magnitude and phase responses determine the steady-state

response of the filter.
• Rise time, settling time, and overshoot specify the transient
response of the filter in the time domain.
• The magnitude response of an ideal filter is given by |H(ω)|= 1 in
the passband and |H(ω)|= 0 in the stopband.
 Practical Filter Specifications
• must accept a more gradual
roll-off between the passband
and the stopband. A transition
band is introduced to permit
the smooth
• magnitude drop-off between
the passband and the
stopband.
• In addition, the deviation from
|H(ω)|= 1 (0 dB) in the
passband is called magnitude
distortion.
• Passband and stopband deviations may be expressed in decibels.
The peak passband ripple, δp, and the minimum stopband
attenuation, δs, in decibels are given as
 FINITE IMPULSE RESPONSE FILTERS

• FIR filter of length L can be represented by its impulse response h(n),

which has at most L nonzero samples. That is, h(n) = 0 for all n ≥ L.
• Some characteristics of FIR filters
• 1. Because there is no feedback of past output samples the FIR fi
lters are always stable. This inherent stability is also manifested in
the absence of poles in the transfer function
• 2. The filter has finite memory from x(n) to x(n − L + 1),
• 3. The design of linear-phase filters can be guaranteed. In many real-
world applications such as audio signal processing, linear-phase fi
lters are preferred because they avoid phase distortion.
 Cont.

• 4. The finite-precision errors are less severe in FIR filters than in IIR
filters.
• 5. FIR filters can be easily implemented on most DSP processors
such as the Blackfin processor.
• 6. A relatively higher-order FIR filter is required to obtain the same
characteristics as compared with an IIR filter, and this may result in
higher computational cost.
 Characteristics and
Implementation of FIR Filters

• Output of the FIR filter as a convolution sum of the input with the
impulse response of the system.
• convolution of the length M input with the length L impulse
response results in a length L + M − 1 output.
• An FIR filter has linear phase if its coefficients satisfy the following
symmetric condition or the antisymmetric (negative symmetry)
condition .
• There are four types of linear-phase FIR filters,
• Type I—Positive symmetry and L is even.
• Type I I—Positive symmetry and L is odd.
• Type III—Negative symmetry and L is even.
• Type IV—Negative symmetry and L is odd.
 Cont.

• The frequency response of the Type I filter is always zero at the

Nyquist frequency (or fs/2). unsuitable for a high-pass filter.
• Type III and IV filters introduce a 90° phase shift; thus they are
often used for designing Hilbert transformers. The frequency
response is always zero at DC frequency, making them unsuitable
for low-pass filters.
• In addition, Type III response is always zero at the Nyquist
frequency, making it also unsuitable for a high-pass filter.
• The symmetry (or antisymmetry) property of a linear-phase FIR fi
lter can be exploited to reduce the total number of multiplications
almost in half.
 Design of FIR Filters

• The objective of FIR filter design is to determine a set of filter

coefficients {bl, l =0, 1, . . . , L − 1} such that the filter
performance is close to the given specifications.
• The Fourier series method offers a very simple and flexible way of
computing FIR filter coefficients, but it does not allow the designer
to control the filter parameters.
Cont.
Cont.
 INFINITE IMPULSE RESPONSE
FILTERS

• Design of IIR Filters

• design of an analog filter in the s-domain and using mapping
technique to transform it into the z-domain.
• Laplace transform of analog signal x(t)

• z-transform can be viewed as the Laplace transform of the sampled

function x(nT) with the change of variable.
 Cont.

• The portion of the jΩ-axis between Ω=−π/T and Ω=π/T in the s-plane
is mapped onto the unit circle in the z-plane from −π to π. As Ω
varies from 0 to ∞, there are an infinite number of encirclements of
the unit circle in the counterclockwise direction.
• Infinite number of encirclements of the unit circle in the clockwise
direction as Ω varies from 0 to −∞.
• mapping from the s-plane to the z-plane is not one-to-one because
many points in the s-plane are mapped to a single point in the z-
plane.
• Because analog filter design is a mature and well-developed field ,
design of digital IIR filters in the analog domain and then convert the
designed analog filter transfer function H(s) into the digital domain.
• The problem is to determine a digital filter H(z) that will approximate
the performance of the desired analog filter H(s).
• There are two commonly used methods,
• the impulse-invariant and
• the bilinear transform,
• for designing digital IIR filters based on existing analog IIR filters.
 Cont.

• The impulse-invariant method preserves the impulse response of

the original analog filter by digitizing the impulse response of the
analog filter but not its frequency (magnitude) response. Because of
inherent aliasing, this method is inappropriate for high-pass or
bandstop filters.
• The bilinear-transform method yields very efficient filters and is well
suited for the design of frequency-selective filters.
• The bilinear transform is defined as
 Cont.

• jΩ-axis of the s-plane (σ= 0) maps onto the unit circle in the z-
plane. The left (σ< 0) and right (σ> 0) halves of the s-plane map
onto the inside and outside of the unit circle, respectively.
• entire jΩ-axis is compressed into the interval −π/T ≤ω≤π/T in a one-
to-one manner. Each point in the s-plane is uniquely mapped onto
the z-plane.
• The entire band ΩT ≥ 1 is compressed onto π/2 ≤ω≤π. This
frequency compression effect associated with the bilinear transform
is known as frequency warping because of the nonlinearity of the
arctangent function .
• This nonlinear frequency warping phenomenon must be taken into
• Consideration This can be done by prewarping the critical
frequencies and using frequency scaling.
 Structures and Characteristics of IIR Filters

• In DSP implementation, consider the required operations, memory

storage, and finite-word length effects.
• H(z) can be realized in several forms or configurations. high-order IIR
filter is factored into second-order sections and connected in cascade
or parallel to form an overall filter.
• direct-form I, direct-form II, cascade, and parallel realizations.
• cascade realization of an IIR filter assumes that the transfer function
is the product of first-order and/or second-order IIR sections. each
Hk(z) is a first- or second-order IIR filter and K is the total number of
sections.
• transfer function for each section of filter for the first-order filter /
second-order section.
• I/O equations describing the time-domain operations
• signal-flow diagram of the second-order IIR filter
 Cont.

• different ordering and pairing, it is possible to obtain many different

cascade realizations for the same transfer function H(z).
• Ordering means the order of connecting Hk(z), and pairing means
the grouping of poles and zeros of H(z) to form a section.
• Each cascade realization behaves differently from others because
of the finite-word length effects. The best ordering is the one that
generates the minimum overall roundoff noise.
• cascade realization variation of one parameter will affect only
pole(s) Therefore, the cascade realization is less sensitive to
parameter variation.
• realize it using the cascade form in terms of first-order and second-
order sections.
• factoring the numerator and denominator polynomials

• different pairings of poles and zeros, there are different realizations

of H(z). For example,
• The overall transfer function is expressed as The parallel
realizations can be realized with the function residuez .

• The system is stable if and only if all its poles are inside the unit
circle. For the cascade structure stability can be guaranteed if every
filter section Hk (z) defined is stable.
 ADAPTIVE FILTERS

• FIR and IIR filters - characteristics of these filters are time invariant
because they have fixed coefficients.
• these filters cannot be applied for time-varying signals and noises.
• adaptive filters’ coefficients are updated automatically by adaptive
algorithms.
• characteristics of adaptive filters are time varying and can adapt to
an unknown and/or changing environment.
• Coefficients of adaptive filters cannot be determined by filter design
software.
 Structures and Algorithms of Adaptive Filters

• example, a modem needs a channel equalizer for transmitting and

receiving data over telecommunication channels. Because the dial-
up channel has different characteristics on each connection and is
time varying, the channel equalizers must be adaptive.
• consists of two functional blocks—
• a digital filter to perform the desired filtering and
• an adaptive algorithm to automatically adjust the coefficients (or
weights) of that filter.
• The adaptive algorithm adjusts the filter coefficients to minimize a
predetermined cost function that is related to e(n)
• FIR and IIR filters can be used for adaptive filtering.
• FIR filter is always stable and can provide a linear-phase response.
• The poles of the IIR filter may move outside the unit circle during
adaptation resulting in an unstable filter.
• adaptive FIR filters are widely used for real-world applications.
• where the filter coefficients wl (n) are time varying and updated by
an adaptive algorithm.
 Cont.

• The objective of the adaptive algorithm is to update the filter

coefficients to minimize some predetermined performance criterion
(or cost function).
• The most commonly used cost function is based on the mean square
error (MSE)
• The steepest-descent method is an iterative (recursive) technique
that starts from an initial (arbitrary) weight vector. The weight vector
is updated at each iteration in the direction of the negative gradient
of the error surface.
• where μ is a convergence factor (or step size) that controls stability
and the rate of descent. The larger the value of μ, the faster the
speed of convergence.
• ∇ξ(n) denotes the gradient of the error function with respect to w(n),
• lead to the minimum MSE, at which point the weight vector reaches
its optimum value.
• method of steepest descent cannot be used directly because it
requires the exact gradient vector.
• most widely used adaptive FIR filter with the least mean square
(LMS) algorithm, or stochastic gradient algorithm, which uses the
instantaneous squared error, e2(n), to estimate the MSE.
 Cont.

• Adaptive FIR filters using the LMS algorithm are relatively simple to
design and implement.
• well understood with regard to stability, convergence speed,
steady-state performance, and finite-precision effects.
• To effectively use the LMS algorithm, we must determine
parameters L, μ, and w(0), where w(0) is the initial weight vector at
time n = 0.
• convergence of the LMS algorithm must satisfy
• Px denotes the power of x(n)
 Cont.

• upper bound on μ is inversely proportional to L, a smaller μ is used

for large-order filters. In addition, μ is inversely proportional to the
input signal power.
• One effective approach is to normalize μ with respect to the input
signal power Px. The resulting algorithm is called the normalized
LMS algorithm,
• where Pˆx(n) is an estimate of the power of x(n) at time n and β is a
normalized step size that satisfies the criterion 0 <β< 2
 Cont.

• commonly used method to estimate Pˆx(n) sample by sample is

similar to the first-order IIR filter.

• Because it is not desirable that the power estimate Pˆx(n) be zero or

very small, a software constraint is required to ensure that the
normalized step size is bounded even if Pˆx(n) is very small.
• Convergence of the MSE toward its minimum value is a commonly
used performance measurement in adaptive systems because of its
simplicity.
 Design and
Applications of
Adaptive Filters

• MATLAB Filter Design Toolbox provides the function adaptfilt for

implementing adaptive filters. h = adaptfilt.algorithm(. ..);
• supports many adaptive algorithms. LMS-type, recursive least
squares, affine projection, and frequency-domain
• The adaptive filter is able to operate in an unknown environment
and to track time variations of the input signals.
• Linear prediction has been successfully applied to a wide range of
applications such as speech coding and separating signals from
noise.
• digital filter in which the coeffiients wl (n) are updated by the LMS
algorithm.
• effective in practical applications when there is insufficient a priori
knowledge of the signal and noise parameters.
• Adaptive noise cancellation employs an adaptive filter to cancel the
noise components in the primary signal picked up by the primary
sensor.
• primary sensor is placed close to the signal source to pick up the
• desired signal. The reference sensor is placed close to the noise
source to sense the noise only.
• The adaptive filter uses the reference input x(n) to estimate the
noise picked up by the primary sensor.
• The filter output y(n) is then subtracted from the primary signal d(n),
producing e(n) as the desired signal plus reduced noise.
• To apply the adaptive noise cancellation effectively, it is critical
• to avoid the signal components from the signal source being picked
up by the reference sensor.
• This “cross talk” effect will degrade the performance because the
presence of the signal components in the reference signal will
cause the adaptive filter to cancel the desired signal.
• The transmission of high-speed data through a channel is limited by
inter symbol interference caused by distortion in the transmission
channel.
• This problem can be solved by using an adaptive equalizer in the
receiver that counteracts the channel distortion.
• original signal x(n) because it was distorted by the overall channel
transfer function C(z), which includes the transmit filter, the
transmission medium, and the receive filter.
• recover the original signal x(n), we need to process s(n) with the
• equalizer W(z), which is the inverse of the channel’s transfer
function C(z), to compensate for the channel distortion.
Thank You

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