Design of a reusable distributed arithmetic filter and its application to the affine projection algorithm

Digital signal processing (DSP) is widely used in many applications spanning the spectrum from audio processing to image and video processing to radar and sonar processing. At the core of digital signal processing applications is the digital filter which are implemented in two ways, using either finite impulse response (FIR) filters or infinite impulse response (IIR) filters. The primary difference between FIR and IIR is that for FIR filters, the output is dependent only on the inputs, while for IIR filters the output is dependent on the inputs and the previous outputs. FIR filters also do not sur from stability issues stemming from the feedback of the output to the input that aect IIR filters. In this thesis, an architecture for FIR filtering based on distributed arithmetic is presented. The proposed architecture has the ability to implement large FIR filters using minimal hardware and at the same time is able to complete the FIR filtering operation in minimal amount of time and delay when compared to typical FIR filter implementations. The proposed architecture is then used to implement the fast affine projection adaptive algorithm, an algorithm that is typically used with large filter sizes. The fast affine projection algorithm has a high computational burden that limits the throughput, which in turn restricts the number of applications. However, using the proposed FIR filtering architecture, the limitations on throughput are removed. The implementation of the fast affine projection adaptive algorithm using distributed arithmetic is unique to this thesis. The constructed adaptive filter shares all the benefits of the proposed FIR filter: low hardware requirements, high speed, and minimal delay.Ph.D.Committee Chair: Anderson, Dr. David V.; Committee Member: Hasler, Dr. Paul E.; Committee Member: Mooney, Dr. Vincent J.; Committee Member: Taylor, Dr. David G.; Committee Member: Vuduc, Dr. Richar

Multi-Step Knowledge-Aided Iterative ESPRIT for Direction Finding

In this work, we propose a subspace-based algorithm for DOA estimation which iteratively reduces the disturbance factors of the estimated data covariance matrix and incorporates prior knowledge which is gradually obtained on line. An analysis of the MSE of the reshaped data covariance matrix is carried out along with comparisons between computational complexities of the proposed and existing algorithms. Simulations focusing on closely-spaced sources, where they are uncorrelated and correlated, illustrate the improvements achieved.Comment: 7 figures. arXiv admin note: text overlap with arXiv:1703.1052

A study on adaptive filtering for noise and echo cancellation.

The objective of this thesis is to investigate the adaptive filtering technique on the application of noise and echo cancellation. As a relatively new area in Digital Signal Processing (DSP), adaptive filters have gained a lot of popularity in the past several decades due to the advantages that they can deal with time-varying digital system and they do not require a priori knowledge of the statistics of the information to be processed. Adaptive filters have been successfully applied in a great many areas such as communications, speech processing, image processing, and noise/echo cancellation. Since Bernard Widrow and his colleagues introduced adaptive filter in the 1960s, many researchers have been working on noise/echo cancellation by using adaptive filters with different algorithms. Among these algorithms, normalized least mean square (NLMS) provides an efficient and robust approach, in which the model parameters are obtained on the base of mean square error (MSE). The choice of a structure for the adaptive filters also plays an important role on the performance of the algorithm as a whole. For this purpose, two different filter structures: finite impulse response (FIR) filter and infinite impulse response (IIR) filter have been studied. The adaptive processes with two kinds of filter structures and the aforementioned algorithm have been implemented and simulated using Matlab.Dept. of Electrical and Computer Engineering. Paper copy at Leddy Library: Theses & Major Papers - Basement, West Bldg. / Call Number: Thesis2005 .J53. Source: Masters Abstracts International, Volume: 44-01, page: 0472. Thesis (M.A.Sc.)--University of Windsor (Canada), 2005

Theory, design and application of gradient adaptive lattice filters

SIGLELD:D48933/84 / BLDSC - British Library Document Supply CentreGBUnited Kingdo

Measuring the Phase Variation of a DOCSIS 3.1 Full Duplex Channel

Including a Full Duplex option into DOCSIS introduces several problems. One of the more troublesome issues is the presence of a strong self interference signal that leaks from the transmit side to the receive side of a cable node. This self interference is caused by echoes in the channel that translate the forward travelling transmit signals into a reverse travelling signal, as well as, by leakage from the hybrid coupler used to couple the upstream and downstream signals. To suppress this self interference an echo canceller is implemented to remove the unwanted interference from the received signal. Unfortunately with the high rates of data transmission used in modern day CATV networks the echo canceller needs tremendous precision. A major concern in the implementation of Full Duplex into DOCSIS is if the channels used are even very slightly time varying. The echos in such channels change with time and can be difficult for the echo canceller to track. Changes in the response of the channel cause the echo profile of the network to shift and the echo canceler to re-adapt to the new channel response. The issue with this changing response is that it is possible for the channel to change faster than the echo canceller can adapt, resulting in the interference becoming unacceptably high. Since the channel is a physical network of coaxial cables often exposed to the environment, its propagation properties can be affected by wind swaying pole mounted cables, or by rapid heating from the sun, or sudden shifts in the load of the network. With information on how the physical properties of the cable changes, the engineers designing the echo canceller can know how fast the canceller must adapt to changes and also have a better measure of how reliable its echo cancellation will be. In this thesis the stability of the echo profile of the channel is measured. It is shown that the property of the channel with the greatest potential to rapidly change and cause noise after echo cancellation is the phase response of the channel. Due to this, the approach of this thesis is to measure the fluctuations in the phase of the channel response of a CATV network constructed in the lab. To measure the fluctuations in the phase response of the channel, a PLL (Phase Locked Loop) based circuit is designed and built on an FPGA (Field Programmable Gate Array) and connected to a model of a simple CATV network. The PLL circuit used to measure the phase fluctuations of the channel is designed to be able to measure changes occurring faster than 0.1 Hz and with a power higher than $10^{-7} \: V^2$. The circuit is able to capture data from the channel over a period of 90 seconds. Using this phase variation measurement circuit a series of experiments were performed on a model CATV DOCSIS network. It was found that many physical disturbances to the network had the effect of rapidly shifting the phase response of the network. Heating the cables in the network was found to shift the phase response upwards of $20000\:\mu$radians. Flexing the cables in the network was found to have a peak phase variation of $8000\: \mu$radians with similar effects found from walking over cables. Overall, it was clear that physical effects on the network had the propensity to rapidly shift the network response. Any echo canceller that is designed in the future will have to consider these effects when reporting the cancellation that it is able to achieve

Efficient time delay estimation and compensation applied to the cancellation of acoustic echo

The system identification problem is notably dealt with using adaptive filtering approaches. In many applications the unknown system response consists of an initial sequence of zero-valued coefficients that precedes the active part of the response. The presence of these coefficients introduces a flat delay in the incoming signals which can take significantly large values. When most adaptive approaches attempt to model such a system, the presence of flat delay impairs their operation and performance. The approach introduced in this thesis aims to model the flat delay and active part of the unknown system separately. An efficient system for time delay estimation (TDE) is introduced to estimate the flat delay of an unknown system. The estimated delay is then compensated within the adaptive system thus allowing the latter to cover the active part ofthe unknown system. The proposed system is applied to the Acoustic Echo Cancellation (ABC) problem

Acoustic Echo Cancellation and their Application in ADF

In this paper, we present an overview of the principal, structure and the application of the echo cancellation and kind of application to improve the performance of the systems. Echo is a process in which a delayed and distorted version o the original sound or voice signal is reflected back to the source. For the acoustic echo canceller much and more study are required to make the good tracking speed fast and reduce the computational complexity. Due to the increasing the processing requirement, widespread implementation had to wait for advances in LSI, VLSI echo canceller appeared. DOI: 10.17762/ijritcc2321-8169.150513