This document provides an introduction to equalization and summarizes several equalization techniques:
1) Zero forcing equalizers aim to completely eliminate intersymbol interference by inverting the channel response but can amplify noise.
2) The mean square error criterion aims to minimize the error between the received and desired signals when filtered by the equalizer. This can be solved using least squares or adaptive algorithms like LMS.
3) The least mean square algorithm approximates the steepest descent method to iteratively and adaptively update the equalizer filter taps to minimize the mean square error based only on instantaneous measurements. This makes it suitable for time-varying channels.
This document discusses adaptive equalization techniques used in wireless communications. It begins by describing different types of interference such as co-channel, adjacent channel, and inter-symbol interference that affect wireless transmissions. Equalization is introduced as a technique to counter inter-symbol interference by concentrating dispersed symbol energy back into its time interval. Adaptive equalization is specifically discussed as it can track time-varying mobile channel characteristics using algorithms like zero forcing, least mean squares, and recursive least squares. The key components of an adaptive equalizer including its operating modes in training and tracking are also outlined.
The document summarizes key concepts in equalization and diversity techniques used in mobile communication systems. It discusses linear equalizers like transversal filters and lattice filters. Nonlinear equalizers covered include decision feedback equalization (DFE) and maximum likelihood sequence estimation (MLSE). DFE uses a feedforward filter and feedback filter to cancel intersymbol interference. MLSE estimates sequences using a trellis channel model and the Viterbi algorithm. Diversity techniques like spatial, frequency and time diversity are also introduced to mitigate fading effects.
This document provides an overview of equalizer design in digital communication systems. It discusses the need for equalization to address inter-symbol interference caused by channel limitations. It describes two main equalizer designs: zero-forcing equalizers that apply the inverse channel response and minimum mean square error equalizers that minimize the error between the equalized signal and desired signal. It explains how the tap coefficients of these equalizers can be calculated using linear algebra methods like solving sets of equations. The document concludes by noting that equalization is a key technique in modern communications to compensate for channel distortions.
1. Equalizers are used to reduce inter-symbol interference in wireless communication and help reduce bit errors at the receiver.
2. There are two main types of equalizers - linear equalizers and non-linear equalizers. Linear equalizers include zero forcing and MMSE equalizers, while non-linear equalizers include decision feedback equalizers.
3. Adaptive equalizers automatically adapt to changing channel properties over time using algorithms like LMS and RLS to update equalizer coefficients.
the presentation consists of a brief description about ADAPTIVE LINEAR EQUALIZER , its classification and the associated attributes of ZERO FORCING EQUALIZER and MMSE EQUALIZER
The document discusses adaptive channel equalization using neural networks. It provides an overview of neural networks and their application to channel equalization. Specifically, it summarizes various neural network architectures that have been used for equalization, including multilayer perceptrons, functional link artificial neural networks, Chebyshev neural networks, and radial basis function networks. It compares the bit error rate performance of these different neural network equalizers with traditional linear equalizers such as LMS and RLS. Overall, the document finds that neural network equalizers can better handle nonlinear channel distortions compared to linear equalizers and that radial basis function networks provide particularly good performance for channel equalization applications.
The document discusses equalization techniques used to mitigate inter-symbol interference (ISI) in digital communication systems. Equalization aims to remove ISI and noise effects from the channel. It is located at the receiver and uses techniques like linear equalizers, decision feedback equalization, and maximum likelihood sequence estimation to estimate the channel response and minimize the error between transmitted and received symbols while balancing noise. As the wireless channel changes over time, adaptive equalization is used where the equalizer periodically trains and tracks the changing channel response.
The document provides an overview of adaptive filters. It discusses that adaptive filters are digital filters that have self-adjusting characteristics to changes in input signals. They have two main components: a digital filter with adjustable coefficients and an adaptive algorithm. Common adaptive algorithms are LMS and RLS. Adaptive filters are used for applications like noise cancellation, system identification, channel equalization, and signal prediction. The key aspects of adaptive filter theory and algorithms like LMS, RLS, Wiener filters are also covered.
This document discusses adaptive equalization techniques used in wireless communications. It begins by describing different types of interference such as co-channel, adjacent channel, and inter-symbol interference that affect wireless transmissions. Equalization is introduced as a technique to counter inter-symbol interference by concentrating dispersed symbol energy back into its time interval. Adaptive equalization is specifically discussed as it can track time-varying mobile channel characteristics using algorithms like zero forcing, least mean squares, and recursive least squares. The key components of an adaptive equalizer including its operating modes in training and tracking are also outlined.
The document summarizes key concepts in equalization and diversity techniques used in mobile communication systems. It discusses linear equalizers like transversal filters and lattice filters. Nonlinear equalizers covered include decision feedback equalization (DFE) and maximum likelihood sequence estimation (MLSE). DFE uses a feedforward filter and feedback filter to cancel intersymbol interference. MLSE estimates sequences using a trellis channel model and the Viterbi algorithm. Diversity techniques like spatial, frequency and time diversity are also introduced to mitigate fading effects.
This document provides an overview of equalizer design in digital communication systems. It discusses the need for equalization to address inter-symbol interference caused by channel limitations. It describes two main equalizer designs: zero-forcing equalizers that apply the inverse channel response and minimum mean square error equalizers that minimize the error between the equalized signal and desired signal. It explains how the tap coefficients of these equalizers can be calculated using linear algebra methods like solving sets of equations. The document concludes by noting that equalization is a key technique in modern communications to compensate for channel distortions.
1. Equalizers are used to reduce inter-symbol interference in wireless communication and help reduce bit errors at the receiver.
2. There are two main types of equalizers - linear equalizers and non-linear equalizers. Linear equalizers include zero forcing and MMSE equalizers, while non-linear equalizers include decision feedback equalizers.
3. Adaptive equalizers automatically adapt to changing channel properties over time using algorithms like LMS and RLS to update equalizer coefficients.
the presentation consists of a brief description about ADAPTIVE LINEAR EQUALIZER , its classification and the associated attributes of ZERO FORCING EQUALIZER and MMSE EQUALIZER
The document discusses adaptive channel equalization using neural networks. It provides an overview of neural networks and their application to channel equalization. Specifically, it summarizes various neural network architectures that have been used for equalization, including multilayer perceptrons, functional link artificial neural networks, Chebyshev neural networks, and radial basis function networks. It compares the bit error rate performance of these different neural network equalizers with traditional linear equalizers such as LMS and RLS. Overall, the document finds that neural network equalizers can better handle nonlinear channel distortions compared to linear equalizers and that radial basis function networks provide particularly good performance for channel equalization applications.
The document discusses equalization techniques used to mitigate inter-symbol interference (ISI) in digital communication systems. Equalization aims to remove ISI and noise effects from the channel. It is located at the receiver and uses techniques like linear equalizers, decision feedback equalization, and maximum likelihood sequence estimation to estimate the channel response and minimize the error between transmitted and received symbols while balancing noise. As the wireless channel changes over time, adaptive equalization is used where the equalizer periodically trains and tracks the changing channel response.
The document provides an overview of adaptive filters. It discusses that adaptive filters are digital filters that have self-adjusting characteristics to changes in input signals. They have two main components: a digital filter with adjustable coefficients and an adaptive algorithm. Common adaptive algorithms are LMS and RLS. Adaptive filters are used for applications like noise cancellation, system identification, channel equalization, and signal prediction. The key aspects of adaptive filter theory and algorithms like LMS, RLS, Wiener filters are also covered.
DSP_2018_FOEHU - Lec 06 - FIR Filter DesignAmr E. Mohamed
This lecture discusses the design of finite impulse response (FIR) filters. It introduces the window method for FIR filter design, which involves truncating the ideal impulse response with a window function to obtain a causal FIR filter. Common window functions are presented such as rectangular, triangular, Hanning, Hamming, and Blackman windows. These windows trade off main lobe width and side lobe levels. The document provides an example design of a low-pass FIR filter using the Hamming window to meet given passband and stopband specifications.
Diversity Techniques in Wireless CommunicationSahar Foroughi
This document discusses diversity techniques for wireless communication, including cooperative diversity. It begins by introducing wireless systems and the impairments they face like fading. It then covers various diversity techniques like space, frequency, and time diversity that provide multiple transmission paths to reduce fading. Cooperative diversity is described as allowing single-antenna devices to achieve MIMO-like benefits by sharing antennas. The document outlines cooperative transmission protocols and challenges at different network layers in implementing cooperation. In conclusion, diversity techniques improve performance by providing multiple signal replicas to overcome fading, while cooperation enables reliability and throughput gains with challenges to address across protocol layers.
The document discusses speech processing and vocoding. It begins by defining speech and how it is produced, including voiced and unvoiced sounds. It then describes the human speech production system and various speech coding techniques like waveform coding, vocoding, and analysis-by-synthesis coding. Finally, it provides details on the G.729 speech codec, including its operations, process flow, specifications, and how it achieves speech compression to 8 kbps from the original 128 kbps.
Lecture Notes on Adaptive Signal Processing-1.pdfVishalPusadkar1
Adaptive filters are time-variant, nonlinear, and stochastic systems that perform data-driven approximation to minimize an objective function. The chapter discusses adaptive filter applications like system identification, inverse modeling, linear prediction, and noise cancellation. It also covers stochastic signal models, optimum linear filtering techniques like Wiener filtering, and solutions to the Wiener-Hopf equations. Numerical techniques like steepest descent are discussed for minimizing the mean square error function in adaptive filters. Stability and convergence analysis is presented for the steepest descent approach.
This document discusses pulse amplitude modulation (PAM). PAM is a digital modulation technique where the amplitude of pulses is varied to represent data symbols. In PAM, each pulse amplitude corresponds to a data symbol value. The document discusses binary and M-ary PAM schemes. It also covers topics like intersymbol interference, eye diagrams, Nyquist pulse shaping criteria, and raised cosine pulse shaping to minimize intersymbol interference at the receiver. PAM is used to convert discrete amplitude symbols into analog pulses for transmission over a channel, and the receiver demodulates the signal to recover the data symbols.
Gaussian Minimum Shift Keying (GMSK) is a form of continuous-phase frequency shift keying that uses a Gaussian filter to generate a constant envelope signal. It provides better spectral efficiency than MSK through bandwidth reduction while maintaining low intersymbol interference. GMSK is used widely in wireless technologies like GSM and CDPD due to its power efficiency and good bit error rate performance compared to other modulation schemes. While more spectrally efficient than MSK, GMSK also has slightly higher error rates and requires more complex receivers.
These slides deal with the basic problem of channel equalization and exposes the issue related to it and shows how it can be balanced by the usage of effective and robust algorithms.
This document is a thesis submitted by Mohammed Abuibaid to Kocaeli University regarding adaptive beam-forming. It discusses various beam-forming techniques including switched array antennas, DSP-based phase manipulation, and beamforming by precoding. It also covers adaptive beamforming algorithms such as LMS, NLMS, RLS, and CM. Various beam patterns generated by these algorithms are presented. The document motivates the need for adaptive beamforming and 3D beamforming to improve energy efficiency in wireless networks.
This document discusses channel equalization techniques for digital communication systems. It describes four main threats in digital communication channels: inter-symbol interference, multipath propagation, co-channel interference, and noise. It then explains various linear equalization techniques like LMS and NLMS adaptive filters that can be used to mitigate inter-symbol interference. Finally, it discusses the need for non-linear equalizers and how multilayer perceptron neural networks can be used for non-linear channel equalization.
Detection and Binary Decision in AWGN ChannelDrAimalKhan
Thermal noise in communication systems is described by a zero-mean white Gaussian random process with a flat power spectral density (PSD) over all frequencies, making it white noise. The average power of white noise is infinite, so a solution is needed. The optimal receive filter, known as a matched filter, can be used to maximize the signal-to-noise ratio. Intersymbol interference (ISI) occurs when pulses interact with each other, such as through convolution with a low-pass channel filter, reducing performance if not addressed.
This document discusses digital communication systems and provides an overview of several key topics:
- It introduces line coding techniques and their properties.
- It describes the basic digital communication block diagram and advantages of digital transmission.
- It discusses intersymbol interference, equalization techniques like zero-forcing equalization, and eye patterns.
- It provides information on topics like noise immunity, regenerative repeaters, and pulse shaping to eliminate intersymbol interference.
This document discusses vestigial sideband (VSB) modulation. VSB modulation is a compromise between single sideband (SSB) and double sideband suppressed carrier (DSB-SC) modulation that overcomes some of the drawbacks of SSB. In VSB, one sideband and a vestige of the other sideband are transmitted together, requiring less bandwidth than DSB but more than SSB. VSB modulation is commonly used for television signal transmission to reduce the bandwidth requirement compared to DSB.
Orthogonal Frequency Division Multiplexing (OFDM)Gagan Randhawa
The document discusses Orthogonal Frequency Division Multiplexing (OFDM), including its principles, advantages, disadvantages and applications. OFDM divides the available spectrum into multiple orthogonal subcarriers, each modulated with a low data rate stream. This makes OFDM robust to multipath fading and intersymbol interference. While OFDM provides high data rates and spectral efficiency, it suffers from issues like high peak-to-average power ratio and sensitivity to frequency errors. OFDM is used in technologies like WiFi, WiMAX and digital audio/video broadcasting.
This document discusses carrier synchronization techniques in digital communication systems. It begins with an introduction to the need for carrier recovery and symbol synchronization at the receiver. It then covers maximum likelihood estimation of signal parameters including carrier phase. Next, it describes carrier phase estimation using a phase-locked loop and decision-directed loops. It explains how the phase-locked loop works to continuously track and update the carrier phase estimate. Finally, it provides an example of decision-directed carrier phase estimation for a double-sideband suppressed carrier pulse amplitude modulation signal.
The attached narrated power point presentation offers a block level and an elementary level mathematical treatment of optical communication systems employing coherent detection. The material will immensely benefit KTU final year B Tech students who prepare for the subject EC 405, Optical Communications.
This document discusses various channel estimation techniques for OFDM systems. It describes pilot structures like block, comb and lattice types and how they are suited for different channel conditions. It also explains training symbol based channel estimation techniques like LS and MMSE. DFT-based channel estimation aims to improve performance by eliminating noise outside the channel delay. Decision directed channel estimation updates the channel coefficients without pilots by using detected signal feedback.
Concept of Diversity & Fading (wireless communication)Omkar Rane
This document discusses concepts related to fading and diversity in wireless communication systems. It introduces fading as signal variations caused by multipath interference from multiple signal propagation paths. It describes two types of fading: large-scale fading due to path loss and shadowing, and small-scale fading which includes fast fading due to mobility and slow fading due to shadowing. It also discusses different diversity techniques that can be used to combat fading, including space, polarization, frequency and time diversity.
This document provides an overview of signal processing techniques used in wireless systems, including diversity and equalization. It discusses various diversity techniques like spatial, temporal, frequency, angular, and polarization diversity as well as macro and micro diversity. It also explains different types of combining diversity including selection, maximal ratio combining, and equal gain combining. The document concludes with sections on linear equalizers such as zero forcing and MMSE, as well as nonlinear equalizers using algorithms like LMS and RLS.
This document discusses equalization, which refers to an accounting methodology used to ensure incentive fees are fairly allocated among investors in funds that pay performance fees. Equalization is necessary when there is active investing and redemptions to prevent inequities between shareholders. Key terms discussed include high water mark, incentive/performance fees, hurdle rate, and the free ride effect experienced by new investors. The document also examines issues like clawbacks that can occur and different equalization methods used like separate share series and consolidated approaches.
This document discusses different types of equalizers used in audio production including graphic, shelving, and parametric equalizers. Graphic equalizers have a fixed number of frequency bands that can each be boosted or cut using individual gain controls. Shelving equalizers broadly boost or cut the high and low frequencies, while parametric equalizers allow control over the central frequency, gain, and bandwidth (Q-factor) of an adjustable frequency band, providing the most precise equalization capabilities.
DSP_2018_FOEHU - Lec 06 - FIR Filter DesignAmr E. Mohamed
This lecture discusses the design of finite impulse response (FIR) filters. It introduces the window method for FIR filter design, which involves truncating the ideal impulse response with a window function to obtain a causal FIR filter. Common window functions are presented such as rectangular, triangular, Hanning, Hamming, and Blackman windows. These windows trade off main lobe width and side lobe levels. The document provides an example design of a low-pass FIR filter using the Hamming window to meet given passband and stopband specifications.
Diversity Techniques in Wireless CommunicationSahar Foroughi
This document discusses diversity techniques for wireless communication, including cooperative diversity. It begins by introducing wireless systems and the impairments they face like fading. It then covers various diversity techniques like space, frequency, and time diversity that provide multiple transmission paths to reduce fading. Cooperative diversity is described as allowing single-antenna devices to achieve MIMO-like benefits by sharing antennas. The document outlines cooperative transmission protocols and challenges at different network layers in implementing cooperation. In conclusion, diversity techniques improve performance by providing multiple signal replicas to overcome fading, while cooperation enables reliability and throughput gains with challenges to address across protocol layers.
The document discusses speech processing and vocoding. It begins by defining speech and how it is produced, including voiced and unvoiced sounds. It then describes the human speech production system and various speech coding techniques like waveform coding, vocoding, and analysis-by-synthesis coding. Finally, it provides details on the G.729 speech codec, including its operations, process flow, specifications, and how it achieves speech compression to 8 kbps from the original 128 kbps.
Lecture Notes on Adaptive Signal Processing-1.pdfVishalPusadkar1
Adaptive filters are time-variant, nonlinear, and stochastic systems that perform data-driven approximation to minimize an objective function. The chapter discusses adaptive filter applications like system identification, inverse modeling, linear prediction, and noise cancellation. It also covers stochastic signal models, optimum linear filtering techniques like Wiener filtering, and solutions to the Wiener-Hopf equations. Numerical techniques like steepest descent are discussed for minimizing the mean square error function in adaptive filters. Stability and convergence analysis is presented for the steepest descent approach.
This document discusses pulse amplitude modulation (PAM). PAM is a digital modulation technique where the amplitude of pulses is varied to represent data symbols. In PAM, each pulse amplitude corresponds to a data symbol value. The document discusses binary and M-ary PAM schemes. It also covers topics like intersymbol interference, eye diagrams, Nyquist pulse shaping criteria, and raised cosine pulse shaping to minimize intersymbol interference at the receiver. PAM is used to convert discrete amplitude symbols into analog pulses for transmission over a channel, and the receiver demodulates the signal to recover the data symbols.
Gaussian Minimum Shift Keying (GMSK) is a form of continuous-phase frequency shift keying that uses a Gaussian filter to generate a constant envelope signal. It provides better spectral efficiency than MSK through bandwidth reduction while maintaining low intersymbol interference. GMSK is used widely in wireless technologies like GSM and CDPD due to its power efficiency and good bit error rate performance compared to other modulation schemes. While more spectrally efficient than MSK, GMSK also has slightly higher error rates and requires more complex receivers.
These slides deal with the basic problem of channel equalization and exposes the issue related to it and shows how it can be balanced by the usage of effective and robust algorithms.
This document is a thesis submitted by Mohammed Abuibaid to Kocaeli University regarding adaptive beam-forming. It discusses various beam-forming techniques including switched array antennas, DSP-based phase manipulation, and beamforming by precoding. It also covers adaptive beamforming algorithms such as LMS, NLMS, RLS, and CM. Various beam patterns generated by these algorithms are presented. The document motivates the need for adaptive beamforming and 3D beamforming to improve energy efficiency in wireless networks.
This document discusses channel equalization techniques for digital communication systems. It describes four main threats in digital communication channels: inter-symbol interference, multipath propagation, co-channel interference, and noise. It then explains various linear equalization techniques like LMS and NLMS adaptive filters that can be used to mitigate inter-symbol interference. Finally, it discusses the need for non-linear equalizers and how multilayer perceptron neural networks can be used for non-linear channel equalization.
Detection and Binary Decision in AWGN ChannelDrAimalKhan
Thermal noise in communication systems is described by a zero-mean white Gaussian random process with a flat power spectral density (PSD) over all frequencies, making it white noise. The average power of white noise is infinite, so a solution is needed. The optimal receive filter, known as a matched filter, can be used to maximize the signal-to-noise ratio. Intersymbol interference (ISI) occurs when pulses interact with each other, such as through convolution with a low-pass channel filter, reducing performance if not addressed.
This document discusses digital communication systems and provides an overview of several key topics:
- It introduces line coding techniques and their properties.
- It describes the basic digital communication block diagram and advantages of digital transmission.
- It discusses intersymbol interference, equalization techniques like zero-forcing equalization, and eye patterns.
- It provides information on topics like noise immunity, regenerative repeaters, and pulse shaping to eliminate intersymbol interference.
This document discusses vestigial sideband (VSB) modulation. VSB modulation is a compromise between single sideband (SSB) and double sideband suppressed carrier (DSB-SC) modulation that overcomes some of the drawbacks of SSB. In VSB, one sideband and a vestige of the other sideband are transmitted together, requiring less bandwidth than DSB but more than SSB. VSB modulation is commonly used for television signal transmission to reduce the bandwidth requirement compared to DSB.
Orthogonal Frequency Division Multiplexing (OFDM)Gagan Randhawa
The document discusses Orthogonal Frequency Division Multiplexing (OFDM), including its principles, advantages, disadvantages and applications. OFDM divides the available spectrum into multiple orthogonal subcarriers, each modulated with a low data rate stream. This makes OFDM robust to multipath fading and intersymbol interference. While OFDM provides high data rates and spectral efficiency, it suffers from issues like high peak-to-average power ratio and sensitivity to frequency errors. OFDM is used in technologies like WiFi, WiMAX and digital audio/video broadcasting.
This document discusses carrier synchronization techniques in digital communication systems. It begins with an introduction to the need for carrier recovery and symbol synchronization at the receiver. It then covers maximum likelihood estimation of signal parameters including carrier phase. Next, it describes carrier phase estimation using a phase-locked loop and decision-directed loops. It explains how the phase-locked loop works to continuously track and update the carrier phase estimate. Finally, it provides an example of decision-directed carrier phase estimation for a double-sideband suppressed carrier pulse amplitude modulation signal.
The attached narrated power point presentation offers a block level and an elementary level mathematical treatment of optical communication systems employing coherent detection. The material will immensely benefit KTU final year B Tech students who prepare for the subject EC 405, Optical Communications.
This document discusses various channel estimation techniques for OFDM systems. It describes pilot structures like block, comb and lattice types and how they are suited for different channel conditions. It also explains training symbol based channel estimation techniques like LS and MMSE. DFT-based channel estimation aims to improve performance by eliminating noise outside the channel delay. Decision directed channel estimation updates the channel coefficients without pilots by using detected signal feedback.
Concept of Diversity & Fading (wireless communication)Omkar Rane
This document discusses concepts related to fading and diversity in wireless communication systems. It introduces fading as signal variations caused by multipath interference from multiple signal propagation paths. It describes two types of fading: large-scale fading due to path loss and shadowing, and small-scale fading which includes fast fading due to mobility and slow fading due to shadowing. It also discusses different diversity techniques that can be used to combat fading, including space, polarization, frequency and time diversity.
This document provides an overview of signal processing techniques used in wireless systems, including diversity and equalization. It discusses various diversity techniques like spatial, temporal, frequency, angular, and polarization diversity as well as macro and micro diversity. It also explains different types of combining diversity including selection, maximal ratio combining, and equal gain combining. The document concludes with sections on linear equalizers such as zero forcing and MMSE, as well as nonlinear equalizers using algorithms like LMS and RLS.
This document discusses equalization, which refers to an accounting methodology used to ensure incentive fees are fairly allocated among investors in funds that pay performance fees. Equalization is necessary when there is active investing and redemptions to prevent inequities between shareholders. Key terms discussed include high water mark, incentive/performance fees, hurdle rate, and the free ride effect experienced by new investors. The document also examines issues like clawbacks that can occur and different equalization methods used like separate share series and consolidated approaches.
This document discusses different types of equalizers used in audio production including graphic, shelving, and parametric equalizers. Graphic equalizers have a fixed number of frequency bands that can each be boosted or cut using individual gain controls. Shelving equalizers broadly boost or cut the high and low frequencies, while parametric equalizers allow control over the central frequency, gain, and bandwidth (Q-factor) of an adjustable frequency band, providing the most precise equalization capabilities.
This document discusses various techniques used to improve mobile radio link performance including equalization, diversity, and channel coding. It describes equalization techniques that compensate for intersymbol interference caused by multipath. It explains different types of diversity including spatial, time, and frequency diversity that are used to mitigate fading. Specifically, it outlines four common spatial diversity techniques: selection diversity, maximal ratio combining, equal gain diversity, and scanning diversity. The document also discusses time diversity and RAKE receivers used in code division multiple access systems to exploit multipath for additional time diversity gain.
This document discusses turbo equalization, which is an iterative signal processing technique that improves data reliability over intersymbol interference (ISI) channels. It works by exchanging soft information between a channel equalizer and decoder in an iterative fashion. This allows both components to utilize reliability data from each other to gradually improve their estimates. The key advantages are improved bit error rate performance close to non-ISI channels with moderate complexity. Applications include magnetic recording, wireless communications, and optical fiber communications where ISI is a problem.
Equalization is a technique, which is introduced to remove interference after received the signal. This works on receiver side. This is like the extension of simple Transmission system...
Lecture note of Industrial Waste Treatment (Elective -III) as per syllabus of Solapur university for BE Civil
Prepared by
Prof S S Jahagirdar,
Associate Professor,
N K ORchid College of Engg and Tech,
Solapur
Turbo equalization is a technique used in digital communication receivers to mitigate inter-symbol interference (ISI) caused by frequency-selective fading channels. It works by iteratively exchanging soft information between a equalizer and decoder, treating the channel as a convolutional code. This iterative process allows both components to continuously improve their estimates. The document discusses the components of a communication system using turbo equalization, including the encoder, interleaver, mapper and channel. It also explains how turbo equalization can provide performance improvements over standard equalization for modulations like QAM in scenarios like mobile communications.
This document discusses different types of equalizers used to reduce inter-symbol interference in communication channels. It describes linear equalizers, which do not use feedback, and nonlinear equalizers, which use feedback of the output signal. Specifically, it outlines decision feedback equalizers (DFE), which use previous decision outputs to estimate and subtract interference on current symbols. Predictive DFE is also discussed, which consists of a feedforward filter and feedback noise predictor. The document compares conventional DFE and predictive DFE, noting predictive DFE is suboptimal due to separate optimization of its filters.
The document discusses adaptive linear equalizers and turbo equalizers. It provides an overview of how adaptive linear equalizers work to compensate for inter-symbol interference caused by time-variant channels. It also describes how turbo equalizers use feedback between an equalizer and decoder to iteratively improve signal estimation. Key components of the receiver like encoders, interleavers, mappers, and the forward-backward algorithm are explained. Applications of turbo equalization in technologies like SC-FDMA, GSM, and packet data transmission are also mentioned.
This document provides an introduction and overview of industrial wastewater treatment. It discusses how industries use water for manufacturing and processing purposes, which becomes wastewater that must be treated before discharge to prevent environmental pollution. The document then outlines some key contaminants found in wastewater and characteristics of industrial wastewater. It describes common wastewater treatment methods including physical, mechanical, chemical and biological processes and provides details on specific unit operations like screening, sedimentation, flotation and biological treatment methods.
Equalization & Channel Estimation of Block & Comb Type CodesAM Publications
Multi-carrier code division multiple access is an attractive choice for high speed wireless communication as it mitigates
the problem of inter symbol interference and also exploits frequency diversity. The work described in this paper is my effort in this
direction. In this paper working of Transmitter and Receiver model of MCCDMA system is presented. We evaluated interference
and bit error rate for multicarrier code division multiple access wireless communication system. In this thesis my concern is find
out the effect of interference in MC-CDMA system. We find out the effect of number of users and signal power on different
parameters for MC-CDMA system. Simulations are given to support the system and receiver design. All the simulation is carried out on MATLAB tool.
Blind Channel Shortening for MIMO-OFDM System Using Zero Padding and Eigen De...ijsrd.com
This paper deals with multiple-input multiple-output (MIMO) broadband wireless communication systems, employing orthogonal frequency-division multiplexing (OFDM). In order to exploit the benefits of OFDM in highly frequency-selective channels, without any significant increase in receiver complexity, a channel shortening prefilter is inserted at the receiver. The main aim of inserting channel shorteners is to shorten the channel so that the main energy of the composite channel is concentrated within a duration smaller than the guard interval inserted while transmission. Thus by including channel shortening equalizers at the receiver the inter symbol interference or the inter block interference can be suppressed. The new approach proposed in this thesis is zero padding approach with Eigen decomposition approach. The advantages of the proposed approaches include immunity to delay spread, resistance to frequency selective fading and simple equalization. This shortening design is a blind one, i.e., a priori knowledge of the MIMO channel impulse response to be shortened is not required, and can be carried out in closed-form.
This document provides a summary of Ben Jabeur Taoufik's background and qualifications. He received a PhD in Signal Processing from Paris Descartes University in 2009, with a thesis on channel shortening techniques in OFDM systems. Currently he works as a Research Scientist at Qatar Mobility Innovations Center, where his work involves signal and image processing applications such as passive RFID systems and analysis of physiological signals. He has published over 10 journal and conference papers in these areas.
The document discusses turbo equalization, which is a receiver technique used to mitigate inter-symbol interference in digital communication systems. It works by formulating the channel equalization problem as a turbo decoding problem, where the channel acts as a convolutional code and error correction coding acts as the second code. The turbo equalizer uses iterative exchange of soft information between the equalizer and decoder to jointly estimate transmitted symbols and bits. This iterative process allows both components to improve their estimates in each round and help achieve better performance than traditional equalizers.
Este documento resume el proceso de desmontaje y reparación de un amortiguador Equalizer II. Explica que el problema común es la entrada de aceite en la cámara negativa debido al desgaste de la junta del pistón principal. Detalla los pasos para desmontarlo, cambiar las juntas dañadas y volver a llenarlo con el aceite correcto. El autor concluye que con un mantenimiento adecuado, el Equalizer es un buen amortiguador y no está mal diseñado. Invita a otros a añadir fotos y
This document discusses the four main classes of amplifiers used in audio equipment: Class A, Class B, Class AB, and Class D. Class A amplifiers are the most accurate but least efficient, conducting electricity continuously. Class B amplifiers use separate circuits for positive and negative halves of the signal. Class AB amplifiers combine aspects of Classes A and B. Class D amplifiers are very efficient but lower fidelity, rapidly switching transistors on and off. The classes balance factors like efficiency, accuracy, cost and heat dissipation.
This document provides a summary of a monograph dedicated to the design of practical coherent, non-coherent and cooperative MIMO-OFDM turbo-transceivers. It introduces MIMO-OFDM and discusses its benefits and applications in standards like LTE, WiFi and WiMAX. It also describes channel estimation and signal detection techniques for MIMO-OFDM systems. The monograph aims to address performance degradation issues that occur under realistic conditions and presents novel iterative signal processing methods for MIMO-OFDM systems.
The document discusses signal statistics and noise. It defines key terms related to signals like continuous versus discrete signals, mean, standard deviation, and signal-to-noise ratio. It also covers topics such as sampling theory, filters, convolution, the discrete Fourier transform and its properties. The DFT decomposes a signal into its frequency components using a sum of sines and cosines, while the inverse DFT performs the synthesis.
This document provides an introduction to digital signal processing. It discusses how signals can be represented digitally by sampling analog signals and converting them to sequences of numbers. This allows signals to be processed using digital processors. Some key benefits of digital signal processing include accuracy, repeatability, flexibility, and easy implementation of nonlinear and time-varying operations in software. The document covers topics such as sampling, analog-to-digital conversion, reconstruction, discrete-time signals and systems, linearity, time-invariance, and examples of basic sequences like sinusoidal, exponential, and geometric sequences.
EC8553 Discrete time signal processing ssuser2797e4
This document contains a 10 question, multiple choice exam on discrete time signal processing. It covers topics like the discrete Fourier transform (DFT), finite word length effects, fixed point vs floating point representation, and FIR filter design. Specifically, it includes questions that calculate the 4 point DFT of a sequence, define twiddle factors, compare DIT and DIF FFT algorithms, and discuss stability and causality of systems.
Digital Signal Processing[ECEG-3171]-Ch1_L03Rediet Moges
This Digital Signal Processing Lecture material is the property of the author (Rediet M.) . It is not for publication,nor is it to be sold or reproduced.
#Africa#Ethiopia
1) The document discusses various topics related to digital communication including sampling theory, analog to digital conversion, pulse code modulation, quantization, coding, and time division multiplexing.
2) In analog to digital conversion, an analog signal is sampled, quantized by assigning it to discrete amplitude levels, and coded by mapping each level to a binary sequence.
3) The Nyquist sampling theorem states that a signal must be sampled at a rate at least twice its highest frequency to avoid aliasing when reconstructing the original signal.
NOISE CANCELLATION USING LMS ALGORITHM
OBJECTIVE
• INTRODUCTION
• ADAPTIVE FILTER
• BLOCK DIAGRAM
• LEAST MEAN SQUARE - LMS
• ADVANTAGES AND DISADVANTAGES
• MATLAB CODE
• CONCLUSION
ADAPTIVE NOISE CANCELLATION
➢ Adaptive noise cancellation is the approach used for estimating a desired
signal d(n) from a noise-corrupted observation.
x(n) = d(n) + v1(n)
➢ Usually the method uses a primary input containing the corrupted signal
and a reference input containing noise correlated in some unknown way
with the primary noise.
➢ The reference input v1(n) can be filtered and subtracted from the primary
input to obtain the signal estimate 𝑑 ̂(n).
➢ As the measurement system is a black box, no reference signal that is
correlated with the noise is available.
An adaptive filter is composed of two parts, the digital filter and the
adaptive algorithm.
• A digital filter with adjustable coefficients wn(z) and an adaptive algorithm
which is used to adjust or modify the coefficients of the filter.
• The adaptive filter can be a Finite Impulse Response FIR filter or an
Infinite Impulse Response IIR filter.
ALGORITHMS FOR ADAPTIVE EQUALIZATION
• There are three different types of adaptive filtering algorithms.
➢ Zero forcing (ZF)
➢ least mean square (LMS)
➢ Recursive least square filter (RLS)
• Recursive least square is an adaptive filter algorithm that recursively finds the coefficients
that minimize a weighted linear least squares cost function relating to the input signals.
• This approach is different from the least mean-square algorithm that aim to reduce the
mean-square error.
Least Mean Square - LMS
• The LMS algorithm in general, consists of two basics procedure:
1. Filtering process, which involve, computing the output (d(n - d)) of a linear filter in
response to the input signal and generating an estimation error by comparing this
output with a desired response as follows:
y(n) is filter output and is the desired response at time n
2. Adaptive process, which involves the automatics adjustment of the parameter of the
filter in accordance with the estimation error.
➢ where wn is the estimate of the weight value vector at time n, x(n) is the input
signal vector.
➢ e(n) is the filter error vector and μ is the step-size, which determines the filter
convergence rate and overall behavior.
➢ One of the difficulties in the design and implementation of the LMS adaptive
filter is the selection of the step-size μ. This parameter must lie in a specific
range, so that the LMS algorithm converges.
➢ LMS algorithm, aims to reduce the mean-square error.
The convergence characteristics of the LMS adaptive algorithm depends on two
factors: the step-size μ and the eigenvalue spread of the autocorrelation matrix .
The step-size μ must lie in a specific range
where 𝜆𝑚𝑎𝑥 is the largest eigenvalue of the autocorrelation matrix Rx.
• A large value of the step-size μ will lead to a faster convergence but may be less
stable around the minimum value. T
This document discusses three methods for equalization in wideband TDMA systems: linear equalization, decision feedback equalization, and maximum likelihood sequence estimation using the Viterbi algorithm. Linear equalization methods like least mean square aim to minimize intersymbol interference but have limited performance. Decision feedback equalization has better performance than linear equalization by cancelling interference from previously decided symbols. Maximum likelihood sequence estimation using the Viterbi algorithm provides the best performance but highest complexity by estimating the most likely transmitted sequence. The document provides examples of equalizer structures and algorithms like LMS for adjusting filter coefficients to minimize intersymbol interference.
Digital Signal Processing (DSP) from basics introduction to medium level book based on Anna University Syllabus! This is just a share of worthfull book!
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Vidyalankar final-essentials of communication systemsanilkurhekar
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4. TransTrans
filterfilter
channelchannel
ReceiverReceiver
filterfilter
Basic Communication SystemBasic Communication System
HHTT(f)(f) HHRR(f)(f)HHcc(f)(f)
∑ +−−=
k
bdck tnkTtthAtY )()()( 0
The received Signal is the transmitted signal, convolved with the channel
And added with AWGN (Neglecting HTx,HRx)
( ) ( )[ ] ( )tnTAt mb
mK
ckmm
kmhAY 0
+−+= ∑≠
ISI -ISI - IInternter SSymbolymbol
IInterferencenterference
Y(t)
Ak Y(tm)
6. Reasons for ISI
• Channel is band limited in
nature
Physics – e.g. parasitic
capacitance in twisted pairs
– limited frequency response
unlimited time response
•Tx filter might add ISI when
channel spacing is crucial.
• Channel has multi-path
reflections
7. Channel Model
• Channel is unknown
• Channel is usually modeled as Tap-Delay-
Line (FIR)
D D D
h(0) h(1) h(2) h(N-1) h(N)
y(n)
x(n)
+
+
+
+
+
8. Example for Measured Channels
The Variation of the Amplitude of the Channel Taps is Random
(changing Multipath)and usually modeled as Railegh distribution
in Typical Urban Areas
10. Equalizer: equalizes the channel – the
received signal would seen like it
passed a delta response.
))(arg())(arg(
)()(1|)(||)(|
|)(|
1
|)(|
fGfG
tthfGfG
fG
fG
CE
totalCE
C
E
−=
=⇒=⋅⇒= δ
11. Need For Equalization
• Need For Equalization:
– Overcome ISI degradation
• Need For Adaptive Equalization:
– Changing Channel in Time
• => Objective:
Find the Inverse of the Channel Response
to reflect a ‘delta channel to the Rx
*Applications (or standards recommend us the channel
types for the receiver to cope with).
12. Zero forcing equalizers
(according to Peak Distortion Criterion)
Tx Ch Eq
qx
∑−=
±±=
=
=⋅−⋅=
2
2 ...2,1,0
0,1
)()(
n m
m
nmTXCnmTq τ
No ISI
Equalizer taps
is described as matrix)2/( nTmTx −
−−
−−−−
=
)1()5.1()2()5.2()3(
)0()5.0()1()5.1()2(
)1()5.0()0()5.0()1(
)2()5.1()1()5.0()0(
TxTxTxTxTx
xTxTxTxTx
TxTxxTxTx
TxTxTxTxx
X
Example: 5tap-Equalizer, 2/T sample rate:
:Force
14. MSE Criterion
2
1
0
])[][(][ nhnxJ
N
n
θθ −= ∑
−
=
Mean Square Error between the received signal
and the desired signal, filtered by the equalizer filter
LS Algorithm LMS Algorithm
Desired Signal
UnKnown Parameter
)Equalizer filter response(
Received Signal
15. LS
• Least Square Method:
– Unbiased estimator
– Exhibits minimum variance )optimal(
– No probabilistic assumptions )only signal
model(
– Presented by Guass )1795( in studies of
planetary motions(
16. LS - Theory
][][][ mmnhns θ∑ −=
Hns θ=][
2
1
0
])[][(][ nhnxJ
N
n
θθ −= ∑
−
=
∑
∑
−
=
−
=
= 1
0
2
1
0
][
][][
ˆ
N
n
N
n
nh
nhnx
θ
Derivative according to: θ
1.
2.
3.
4.
MSE:
17. The minimum LS error would be obtained by substituting 4 to 3:
][]
])[][(
][
][][ˆ][
])[ˆ][(][ˆ])[ˆ][(][
])[ˆ][])([ˆ][(])[ˆ][(][
2
1
0
2
1
0
1
0
2
min
1
0
1
0
2
)ˆ(0
1
0
1
0
1
0
2
1
0
min
nh
nhnx
nxJ
nhnxnx
nhnxnhnhnxnx
nhnxnhnxnhnxJJ
N
n
N
n
N
n
N
n
N
n
tinhBySubstitu
N
n
N
n
N
n
N
n
∑
∑
∑
∑∑
∑∑
∑∑
−
=
−
=
−
=
−
=
−
=
−
=
−
=
−
=
−
=
−==>
−=
−−−=
−−=−==
θ
θθθ
θθθθ
θ
Energy Of
Original Signal
Energy Of
Fitted Signal
][][ nwSignalnx += If Noise Small enough )SNR large enough(: Jmin~0
Back-Up
18. Finding the LS solution
θHns =][
])][(])][(
])[ˆ][])([ˆ][(])[ˆ][(][
1
0
2
1
0
θθ
θθθθ
HnxHnx
nhnxnhnxnhnxJ
T
N
n
N
n
−−=
−−=−= ∑∑
−
=
−
=
θθθ
θθθθθ
HHHxzx
HHxHHxxxJ
TT
scalar
TT
TTTTTT
+−=
+−−=
2
][
)H: observation matrix )Nxp( and
θ
θ
θ
HHxH
J T
scalar
T
22
)(
+−=
∂
∂
xHHH TT 1
)(ˆ −
=θ
T
Nsssns ])1[],...1[],0[(][ −=
19. LS : Pros & Cons
•Advantages:
•Optimal approximation for the Channel- once calculated
it could feed the Equalizer taps.
•Disadvantages:
•heavy Processing )due to matrix inversion which by
It self is a challenge(
•Not adaptive )calculated every once in a while and
is not good for fast varying channels
• Adaptive Equalizer is required when the Channel is time variant
)changes in time( in order to adjust the equalizer filter tap
Weights according to the instantaneous channel properties.
21. SYSTEM BLOCK USING THE LMS
U[n] = Input signal from the channel ; d[n] = Desired Response
H[n] = Some training sequence generator
e[n] = Error feedback between :
A.) desired response.
B.) Equalizer FIR filter output
W = Fir filter using tap weights vector
22. STEEPEST DESCENT METHOD
• Steepest decent algorithm is a gradient based method which
employs recursive solution over problem )cost function(
• The current equalizer taps vector is W)n( and the next
sample equalizer taps vector weight is W)n+1(, We could
estimate the W)n+1( vector by this approximation:
• The gradient is a vector pointing in the direction of the
change in filter coefficients that will cause the greatest
increase in the error signal. Because the goal is to minimize
the error, however, the filter coefficients updated in the
direction opposite the gradient; that is why the gradient term
is negated.
• The constant μ is a step-size. After repeatedly adjusting
each coefficient in the direction opposite to the gradient of
the error, the adaptive filter should converge.
])[(5.0]1[][ nJnWnW −∇++= µ
23. • Given the following function we need to obtain the vector
that would give us the absolute minimum.
• It is obvious that
give us the minimum.
STEEPEST DESCENT EXAMPLE
2
2
2
121 ),( CCccY +=
,021 == CC
1C
2C
y
Now lets find the solution by the steepest descend method
24. • We start by assuming (C1 = 5, C2 = 7)
• We select the constant . If it is too big, we miss the
minimum. If it is too small, it would take us a lot of time to
het the minimum. I would select = 0.1.
• The gradient vector is:
STEEPEST DESCENT EXAMPLE
µ
µ
][2
1
][2
1
][2
1
][2
1
]1[2
1
9.01.02.0
nnnnn
C
C
C
C
C
C
y
C
C
C
C
=
−
=∇∗−
=
+
=
=∇
2
1
2
1
2
2
C
C
dc
dy
dc
dy
y
• So our iterative equation is:
26. MMSE CRITERIA FOR THE LMS
• MMSE – Minimum mean square error
• MSE =
• To obtain the LMS MMSE we should derivative
the MSE and compare it to 0:
•
}])()()({[(})]()({[( 22
∑−=
−−=−
N
Nn
nkunwkdEkykdE
)(
))()()()()(2})({(
)(
)(
2
kdW
mnRmwnwnPnwkdEd
kdW
MSEd
N
Nn
N
Nm
N
Nn
du ∑ ∑∑ −= −=−=
−+−
=
)}()({)(
)}()({)(
)()()()()(2})({}])()()({[( 22
knukmuEmnR
knukdEnP
mnRmwnwnPnwkdEnkunwkdE
uu
du
N
Nn
N
Nm
N
Nn
du
N
Nn
−−=−
−=
−+−=−− ∑ ∑∑∑ −= −=−=−=
27. MMSE CRITERION FOR THE LMS
,...2,1,0),(][2)(2
)(
)(
)( ±±=−+−==∇ ∑−−
kknRnwkP
kdW
MSEd
nJ uu
N
Nn
du
And finally we get:
By comparing the derivative to zero we get the MMSE:
PRwopt •= −1
This calculation is complicated for the DSP (calculating the inverse
matrix ), and can cause the system to not being stable cause if there
are NULLs in the noise, we could get very large values in the inverse
matrix. Also we could not always know the Auto correlation matrix of the
input and the cross-correlation vector, so we would like to make an
approximation of this.
28. LMS – APPROXIMATION OF THE
STEEPEST DESCENT METHOD
W(n+1) = W(n) + 2*[P – Rw(n)] <= According the MMSE criterion
We assume the following assumptions:
• Input vectors :u(n), u(n-1),…,u(1) statistically independent vectors.
• Input vector u(n) and desired response d(n), are statistically independent of
d(n), d(n-1),…,d(1)
• Input vector u(n) and desired response d(n) are Gaussian-distributed R.V.
•Environment is wide-sense stationary;
In LMS, the following estimates are used:
Ruu^ = u(n)u
H
(n) – Autocorrelation matrix of input signal
Pud^ = u(n)d*(n) - Cross-correlation vector between U[n] and d[n].
*** Or we could calculate the gradient of |e[n]|2
instead of E{|e[n]|2
}
30. LMS STABILITY
The size of the step size determines the algorithm convergence
rate. Too small step size will make the algorithm take a lot of
iterations. Too big step size will not convergence the weight taps.
Rule Of Thumb:
RPN )12(5
1
+
=µ
Where, N is the equalizer length
Pr, is the received power (signal+noise)
that could be estimated in the receiver.
31. LMS – CONVERGENCE GRAPH
This graph illustrates the LMS algorithm. First we start from guessing
the TAP weights. Then we start going in opposite the gradient vector,
to calculate the next taps, and so on, until we get the MMSE,
meaning the MSE is 0 or a very close value to it.(In practice we can
not get exactly error of 0 because the noise is a random process, we
could only decrease the error below a desired minimum)
Example for the Unknown Channel of 2nd
order:
Desired Combination of tapsDesired Combination of taps
33. LMS – EQUALIZER EXAMPLE
Channel equalization
example:
Average Square Error as a
function of iterations number
using different channel
transfer function
(change of W)
34.
35. LMS – Advantage:
• Simplicity of implementation
• Not neglecting the noise like Zero forcing equalizer
• By pass the need for calculating an inverse matrix.
LMS : Pros & Cons
LMS – Disadvantage:
Slow Convergence
Demands using of training sequence as reference
,thus decreasing the communication BW.
36. Non linear equalization
Linear equalization (reminder):
• Tap delayed equalization
• Output is linear combination of the equalizer
input
C
E
G
G
1
=
...
)(
)(
)(
2
3
1
10
1
+++==
−=
−−
−
∏
zazaaC
zX
zY
zaG
E
i
iE as FIR
...)2()1()()( 210 +−⋅+−⋅+⋅= nxanxanxany
37. Non linear equalization – DFE
(Decision feedback Equalization)
∑ ∑ −⋅−−⋅= )()()( inybinxany ii
Advantages: copes with larger ISI
∏
∏
−
−
−
−
==
i
i
i
i
E
zb
za
G
zX
zY
)(
)(
)(
)(
1
1
as IIR
A(z)
Receiver
detector
B(z)
+In Output
+
-
The nonlinearity is due the
detector characteristics that
is fed back (MAPPER)
The Decision feedback leads poles in z domain
Disadvantages: instability danger
39. Blind Equalization
• ZFE and MSE equalizers assume
option of training sequence for
learning the channel.
• What happens when there is
none?
– Blind Equalization
Adaptive
Equalizer
Decision
+
-
Input Output
Error
Signal
nV
nIˆ
nI
~
nd
neBut Usually employs also :
InterleavingDeInterleaving
Advanced coding
ML criterion
Why? Blind Eq is hard and complicated enough!
So if you are going to implement it, use the best blocks
For decision (detection) and equalizing
With LMS
40. Turbo Equalization
MAP
Decoder
+ Π
Π
−1
MAP
Equalizer
+
Channel
Estimator
r
L
D
e(c’) L
D
e(c) L
D
(c)
L
D
(d)
L
E
e(c)L
E
e(c’)L
E
(c’)
Iterative :
Estimate
Equalize
Decode
ReEncode
Usually employs also :
InterleavingDeInterleaving
TurboCoding (Advanced iterative code)
MAP (based on ML criterion)
Why? It is complicated enough!
So if you are going to implement it, use the best blocks
Next iteration would rely on better estimation
therefore would lead more precise equalization
42. ML criterion
• MSE optimizes detection up to 1st
/2nd
order
statistics.
• In Uri’s Class:
– Optimum Detection:
• Strongest Survivor
• Correlation (MF)
(allow optimal performance for Delta ch and Additive noise.
Optimized Detection maximizes prob of detection
(minimizes error or Euclidean distance in Signal Space)
• Lets find the Optimal Detection Criterion while in
presence of memory channel (ISI)
43. ML criterion –Cont.
• Maximum Likelihood :
Maximizes decision probability for the received trellis
Example BPSK (NRZI)
bESS =−= 01
2possible transmitted signals
Energy Per Bit
kbk nEr +±=
Received Signal occupies AWGN
−
−= 2
2
1
2
)(
exp
2
1
)|(
n
bk
n
k
Er
srp
σπσ
+
−= 2
2
0
2
)(
exp
2
1
)|(
n
bk
n
k
Er
srp
σπσ
Conditional PDF (prob of correct decision on r1 pending s1 was transmitted…)
N0/2
Prob of correct decision on a sequence of symbols
∏=
=
K
k
m
kk
m
k srpsrrrp
1
)()(
21 )|()|,...,,(
Transmitted sequence
optimal
44. ML – Cont.
With logarithm operation, it could be shown that this is equivalent to:
Minimizing the Euclidean distance metric of the sequence:
∑=
−=
K
k
m
kk
m
srsrD
1
2)()(
)(),(
How could this be used?
Looks Similar?
while MSE minimizes Error (maximizes Prob) for decision on certain Sym,
MLSE minimizes Error (maximizes Prob) for decision on certain Trellis ofSym,
(Called Metric)
46. We Always disqualify one metric for possible S0 and possible S1.
Finally we are left with 2 options for possible Trellis.
Finally are decide on the correct Trellis with the Euclidean
Metric of each or with Apost DATA
47.
48. References
• John G.Proakis – Digital Communications.
• John G.Proakis –Communication Systems Eng.
• Simon Haykin - Adaptive Filter Theory
• K Hooli – Adaptive filters and LMS
• S.Kay – Statistical Signal Processing – Estimation Theory