Impulse Response Measurement and Equalization Digital Signal Processing LPP Erasmus Program Aveiro 2012 Digital Signal Processing LPP Erasmus Program Aveiro.

Slides:



Advertisements
Similar presentations
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: The Linear Prediction Model The Autocorrelation Method Levinson and Durbin.
Advertisements

1. INTRODUCTION In order to transmit digital information over * bandpass channels, we have to transfer the information to a carrier wave of.appropriate.
Adaptive Filters S.B.Rabet In the Name of GOD Class Presentation For The Course : Custom Implementation of DSP Systems University of Tehran 2010 Pages.
Digital signal processing -G Ravi kishore. INTRODUCTION The goal of DSP is usually to measure, filter and/or compress continuous real-world analog signals.
OPTIMUM FILTERING.
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: The FIR Adaptive Filter The LMS Adaptive Filter Stability and Convergence.
ELE Adaptive Signal Processing
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: Newton’s Method Application to LMS Recursive Least Squares Exponentially-Weighted.
Simple Neural Nets For Pattern Classification
280 SYSTEM IDENTIFICATION The System Identification Problem is to estimate a model of a system based on input-output data. Basic Configuration continuous.
Discrete-Time Signals and Systems Quote of the Day Mathematics is the tool specially suited for dealing with abstract concepts of any kind and there is.
Goals of Adaptive Signal Processing Design algorithms that learn from training data Algorithms must have good properties: attain good solutions, simple.
MM3FC Mathematical Modeling 3 LECTURE 2 Times Weeks 7,8 & 9. Lectures : Mon,Tues,Wed 10-11am, Rm.1439 Tutorials : Thurs, 10am, Rm. ULT. Clinics : Fri,
EECS 20 Chapter 9 Part 21 Convolution, Impulse Response, Filters Last time we Revisited the impulse function and impulse response Defined the impulse (Dirac.
Matched Filters By: Andy Wang.
EE513 Audio Signals and Systems Wiener Inverse Filter Kevin D. Donohue Electrical and Computer Engineering University of Kentucky.
Spectral Analysis Spectral analysis is concerned with the determination of the energy or power spectrum of a continuous-time signal It is assumed that.
Adaptive Signal Processing
Normalised Least Mean-Square Adaptive Filtering
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: Adaptive Noise Cancellation ANC W/O External Reference Adaptive Line Enhancement.
RLSELE Adaptive Signal Processing 1 Recursive Least-Squares (RLS) Adaptive Filters.
Chapter 5ELE Adaptive Signal Processing 1 Least Mean-Square Adaptive Filtering.
Digital Communications Fredrik Rusek Chapter 10, adaptive equalization and more Proakis-Salehi.
Digital Signals and Systems
Dept. of EE, NDHU 1 Chapter Three Baseband Demodulation/Detection.
Discrete-Time and System (A Review)
1 Chapter 8 The Discrete Fourier Transform 2 Introduction  In Chapters 2 and 3 we discussed the representation of sequences and LTI systems in terms.
Equalization in a wideband TDMA system
1 Signals & Systems Spring 2009 Week 3 Instructor: Mariam Shafqat UET Taxila.
Chapter 2: Discrete time signals and systems
Introduction to Adaptive Digital Filters Algorithms
Dept. of EE, NDHU 1 Chapter Three Baseband Demodulation/Detection.
Time-Domain Representations of LTI Systems
Random Processes ECE460 Spring, Power Spectral Density Generalities : Example: 2.
1 Part 5 Response of Linear Systems 6.Linear Filtering of a Random Signals 7.Power Spectrum Analysis 8.Linear Estimation and Prediction Filters 9.Mean-Square.
Real time DSP Professors: Eng. Julian Bruno Eng. Mariano Llamedo Soria.
Correlated and Uncorrelated Signals Problem: we have two signals and. How “close” are they to each other? Example: in a radar (or sonar) we transmit a.
Jessica Arbona & Christopher Brady Dr. In Soo Ahn & Dr. Yufeng Lu, Advisors.
CHAPTER 4 Adaptive Tapped-delay-line Filters Using the Least Squares Adaptive Filtering.
Unit-V DSP APPLICATIONS. UNIT V -SYLLABUS DSP APPLICATIONS Multirate signal processing: Decimation Interpolation Sampling rate conversion by a rational.
Adv DSP Spring-2015 Lecture#9 Optimum Filters (Ch:7) Wiener Filters.
Course Outline (Tentative) Fundamental Concepts of Signals and Systems Signals Systems Linear Time-Invariant (LTI) Systems Convolution integral and sum.
1 Lecture 1: February 20, 2007 Topic: 1. Discrete-Time Signals and Systems.
ECE 5525 Osama Saraireh Fall 2005 Dr. Veton Kepuska
EE513 Audio Signals and Systems
LEAST MEAN-SQUARE (LMS) ADAPTIVE FILTERING. Steepest Descent The update rule for SD is where or SD is a deterministic algorithm, in the sense that p and.
CHAPTER 5 SIGNAL SPACE ANALYSIS
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: Derivation Computational Simplifications Stability Lattice Structures.
Fourier Analysis of Signals and Systems
CHAPTER 2 Discrete-Time Signals and Systems in the Time-Domain
Chapter 1 Random Process
Professors: Eng. Diego Barral Eng. Mariano Llamedo Soria Julian Bruno
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: Normal Equations The Orthogonality Principle Solution of the Normal Equations.
Course Outline (Tentative) Fundamental Concepts of Signals and Systems Signals Systems Linear Time-Invariant (LTI) Systems Convolution integral and sum.
Chapter 2. Fourier Representation of Signals and Systems
Discrete-time Random Signals
DTFT continue (c.f. Shenoi, 2006)  We have introduced DTFT and showed some of its properties. We will investigate them in more detail by showing the associated.
1 st semester 1436 / Modulation Continuous wave (CW) modulation AM Angle modulation FM PM Pulse Modulation Analog Pulse Modulation PAMPPMPDM Digital.
State-Space Recursive Least Squares with Adaptive Memory College of Electrical & Mechanical Engineering National University of Sciences & Technology (NUST)
Geology 6600/7600 Signal Analysis 26 Oct 2015 © A.R. Lowry 2015 Last time: Wiener Filtering Digital Wiener Filtering seeks to design a filter h for a linear.
Lecture 09b Finite Impulse Response (FIR) Filters
Chapter 2. Signals and Linear Systems
ENEE 322: Continuous-Time Fourier Transform (Chapter 4)
Lect2 Time Domain Analysis
Spectral Analysis Spectral analysis is concerned with the determination of the energy or power spectrum of a continuous-time signal It is assumed that.
Adaptive Filters Common filter design methods assume that the characteristics of the signal remain constant in time. However, when the signal characteristics.
Digital Control Systems (DCS)
Outline Introduction Signal, random variable, random process and spectra Analog modulation Analog to digital conversion Digital transmission through baseband.
Equalization in a wideband TDMA system
Equalization in a wideband TDMA system
Presentation transcript:

Impulse Response Measurement and Equalization Digital Signal Processing LPP Erasmus Program Aveiro 2012 Digital Signal Processing LPP Erasmus Program Aveiro 2012

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response measurement Measurement of impulse responses is a common task in different areas (specially important in audio processing) Typically, PC hardware is used to perform this task using the following setup: The following assumptions are made: The whole system between x[n] and y[n] shows time-invariant behaviour. All involved components are sufficiently linear The impulse response can be considered finite (h[n]≈0 for n > M) h(t) DAC ADC x[n] y[n] x(t) y(t) PC

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response measurement In theory, any input signal can be used to measure the impulse response of a linear system. The output signal can be expressed in terms of this impulse response by: Assuming that the impulse response only has N significant samples, the former equation can be rewritten in matrix form as: where L is the length of the input signal

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response measurement Then, the impulse response can be theoretically obtained as: where X -1* represents the pseudo-inverse of matrix X: (X T ·X) -1 ·X T or X T ·(X·X T ) -1 However, in practice, very long input signals would be needed to attain high suppression of measurement noise, resulting in extremely large problems which may turn out to become computationally intractable.

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response measurement An atractive approach consist in using as input signal one whose auto-correlation function is similar to a delta sequence: In this case, the impulse response of the system can be obtained by cross-correlating the output and the input signals:

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response measurement Assuming that the length of the input signal is K and that the impulse response has only N significant samples, we have:

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response measurement Using this indirect approach, a great inmunity against noise is achieved:  The energy of the input signal can be as large as desired, since we have just to extend its duration.  As noise corrupting the output signal is uncorrelated with the input signal x, its contribution in the measurement of the impulse response is very small:

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response measurement Typical test signals: Chirp (Continuous-time frequency varying sinusoid) PR sequence (13-bit Barker Code) White Gaussian Noise x(t) R xx (  )

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response equalization An Equalizer is a system whose main objective is to undo the effect of a previous stage, typically a transmission channel: Ideally, the composite impulse response of the channel and the equalizer is a delta (flat frequency response), so that the output of the equalizer is a delayed version of the input signal. 1.Fixed equalizers (its coefficients do not change with time) 2.Adaptive equalizers (its coefficients are updated periodically based on the current channel characteristics) We can distinguish between two types of equalizers: + Channel c[n] Equalizer g[n]

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Fixed Equalizer (Zero-Forcing) Impulse response equalization In zero-forcing equalization, the equalizer response g[n] attempts to completely inverse the channel by forcing : Or using matrix representation:

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response equalization Adaptive Equalizer When the channel is time-varying (LTV), it is necessary to update the equalizer coefficients in order to track the channel changes. Define the input signal to the equalizer as a vector c k where: and an equalizer weight vector g k, where Then, the output signal is given by: And the error signal e k is given by:

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response equalization An adaptive channel equalizer has the following structure. Adaptive Equalizer

Impulse response measurement and equalization LPP Erasmus Visit, Aveiro 2012 Impulse response equalization The least mean-square (LMS) algorithm carries out the mean squared error by recursively updating the coefficients using the following rules: The step parameter , controls the adaptation rate and must be chosen carefully to guarantee convergence. The equalizer is converged if the error e k becomes steady. Adaptive Equalizer