Adaptively combined LMS and logistic equalizers

An adaptive, convex linear combination of the outputs of a standard least mean square (LMS) equalizer and a sigmoidal equalizer is proposed. This procedure results in improving the speed of the LMS equalizer while retaining the low steady-state error of the sigmoidal filter. Appropriate adaption schemes for both of the filters and for the combination parameters are established. Simulations of practical communication applications demonstrate the effectiveness of this adaptive combination.Publicad

Adaptation algorithms for data echo cancellation using nonquadratic cost functions

Adaptation algorithms for data echo cancellation using nonquadratic cost function

Available Techniques for Magnetic Hard Disk Drive Read Channel Equalization

This paper presents an extensive, non-exhaustive, study of available hard disk drive read channel equalization techniques used in the storage and readback of magnetically stored information. The physical elements and basic principles of the storage processes are introduced together with the basic theoretical definitions and models. Both read and write processes in magnetic storage are explained along with the definition of simple key concepts such as user bit density, intersymbol interference, linear and areal density, read head pulse response models, and coding algorithm

Design of adaptive analog filters for magnetic front-end read channels

Esta tese estuda o projecto e o comportamento de filtros em tempo contínuo de muito-alta-frequência. A motivação deste trabalho foi a investigação de soluções de filtragem para canais de leitura em sistemas de gravação e reprodução de dados em suporte magnético, com custos e consumo (tamanho total inferior a 1 mm2 e consumo inferior a 1mW/polo), inferiores aos circuitos existentes. Nesse sentido, tal como foi feito neste trabalho, o rápido desenvolvimento das tecnologias de microelectrónica suscitou esforços muito significativos a nível mundial com o objectivo de se investigarem novas técnicas de realização de filtros em circuito integrado monolítico, especialmente em tecnologia CMOS (Complementary Metal Oxide Semiconductor). Apresenta-se um estudo comparativo a diversos níveis hierárquicos do projecto, que conduziu à realização e caracterização de soluções com as características desejadas. Num primeiro nível, este estudo aborda a questão conceptual da gravação e transmissão de sinal bem como a escolha de bons modelos matemáticos para o tratamento da informação e a minimização de erro inerente às aproximações na conformidade aos princípios físicos dos dispositivos caracterizados. O trabalho principal da tese é focado nos níveis hierárquicos da arquitectura do canal de leitura e da realização em circuito integrado do seu bloco principal – o bloco de filtragem. Ao nível da arquitectura do canal de leitura, apresenta-se um estudo alargado sobre as metodologias existentes de adaptação de sinal e recuperação de dados em suporte magnético. Este desígnio aparece no âmbito da proposta de uma solução de baixo custo, baixo consumo, baixa tensão de alimentação e baixa complexidade, alicerçada em tecnologia digital CMOS, para a realização de um sistema DFE (Decision Feedback Equalization) com base na igualização de sinal utilizando filtros integrados analógicos em tempo contínuo. Ao nível do projecto de realização do bloco de filtragem e das técnicas de implementação de filtros e dos seus blocos constituintes em circuito integrado, concluiu-se que a técnica baseada em circuitos de transcondutância e condensadores, também conhecida como filtros gm-C (ou transcondutância-C), é a mais adequada para a realização de filtros adaptativos em muito-alta-frequência. Definiram-se neste nível hierárquico mais baixo, dois subníveis de aprofundamento do estudo no âmbito desta tese, nomeadamente: a pesquisa e análise de estruturas ideais no projecto de filtros recorrendo a representações no espaço de estados; e, o estudo de técnicas de realização em tecnologia digital CMOS de circuitos de transcondutância para a implementação de filtros integrados analógicos em tempo contínuo. Na sequência deste estudo, apresentam-se e comparam-se duas estruturas de filtros no espaço de estados, correspondentes a duas soluções alternativas para a realização de um igualador adaptativo realizado por um filtro contínuo passa-tudo de terceira ordem, para utilização num canal de leitura de dados em suporte magnético. Como parte constituinte destes filtros, apresenta-se uma técnica de realização de circuitos de transcondutância, e de realização de condensadores lineares usando matrizes de transístores MOSFET para processamento de sinal em muito-alta-frequência realizada em circuito integrado usando tecnologia digital CMOS submicrométrica. Apresentam-se métodos de adaptação automática capazes de compensar os erros face aos valores nominais dos componentes, devidos às tolerâncias inerentes ao processo de fabrico, para os quais apresentamos os resultados de simulação e de medição experimental obtidos. Na sequência deste estudo, resultou igualmente a apresentação de um circuito passível de constituir uma solução para o controlo de posicionamento da cabeça de leitura em sistemas de gravação/reprodução de dados em suporte magnético. O bloco proposto é um filtro adaptativo de primeira ordem, com base nos mesmos circuitos de transcondutância e técnicas de igualação propostos e utilizados na implementação do filtro adaptativo de igualação do canal de leitura. Este bloco de filtragem foi projectado e incluído num circuito integrado (Jaguar) de controlo de posicionamento da cabeça de leitura realizado para a empresa ATMEL em Colorado Springs, e incluído num produto comercial em parceria com uma empresa escocesa utilizado em discos rígidos amovíveis.This thesis studies the design and behavior of continuous-time very-high-frequency filters. The motivation of this work was the search for filtering solutions for the readchannel in recording and reproduction of data on magnetic media systems, with costs and consumption (total size less than 1 mm2 and consumption under 1mW/pole), lower than the available circuits. Accordingly, as was done in this work, the rapid development of microelectronics technology raised very significant efforts worldwide in order to investigate new techniques for implementing such filters in monolithic integrated circuit, especially in CMOS technology (Complementary Metal Oxide Semiconductor). We present a comparative study on different hierarchical levels of the project, which led to the realization and characterization of solutions with the desired characteristics. In the first level, this study addresses the conceptual question of recording and transmission of signal and the choice of good mathematical models for the processing of information and minimization of error inherent in the approaches and in accordance with the principles of the characterized physical devices. The main work of this thesis is focused on the hierarchical levels of the architecture of the read channel and the integrated circuit implementation of its main block - the filtering block. At the architecture level of the read channel this work presents a comprehensive study on existing methodologies of adaptation and signal recovery of data on magnetic media. This project appears in the sequence of the proposed solution for a lowcost, low consumption, low voltage, low complexity, using CMOS digital technology for the performance of a DFE (Decision Feedback Equalization) based on the equalization of the signal using integrated analog filters in continuous time. At the project level of implementation of the filtering block and techniques for implementing filters and its building components, it was concluded that the technique based on transconductance circuits and capacitors, also known as gm-C filters is the most appropriate for the implementation of very-high-frequency adaptive filters. We defined in this lower level, two sub-levels of depth study for this thesis, namely: research and analysis of optimal structures for the design of state-space filters, and the study of techniques for the design of transconductance cells in digital CMOS circuits for the implementation of continuous time integrated analog filters. Following this study, we present and compare two filtering structures operating in the space of states, corresponding to two alternatives for achieving a realization of an adaptive equalizer by the use of a continuous-time third order allpass filter, as part of a read-channel for magnetic media devices. As a constituent part of these filters, we present a technique for the realization of transconductance circuits and for the implementation of linear capacitors using arrays of MOSFET transistors for signal processing in very-high-frequency integrated circuits using sub-micrometric CMOS technology. We present methods capable of automatic adjustment and compensation for deviation errors in respect to the nominal values of the components inherent to the tolerances of the fabrication process, for which we present the simulation and experimental measurement results obtained. Also as a result of this study, is the presentation of a circuit that provides a solution for the control of the head positioning on recording/playback systems of data on magnetic media. The proposed block is an adaptive first-order filter, based on the same transconductance circuits and equalization techniques proposed and used in the implementation of the adaptive filter for the equalization of the read channel. This filter was designed and included in an integrated circuit (Jaguar) used to control the positioning of the read-head done for ATMEL company in Colorado Springs, and part of a commercial product used in removable hard drives fabricated in partnership with a Scottish company

An investigation of the utility of monaural sound source separation via nonnegative matrix factorization applied to acoustic echo and reverberation mitigation for hands-free telephony

In this thesis we investigate the applicability and utility of Monaural Sound Source Separation (MSSS) via Nonnegative Matrix Factorization (NMF) for various problems related to audio for hands-free telephony. We first investigate MSSS via NMF as an alternative acoustic echo reduction approach to existing approaches such as Acoustic Echo Cancellation (AEC). To this end, we present the single-channel acoustic echo problem as an MSSS problem, in which the objective is to extract the users signal from a mixture also containing acoustic echo and noise. To perform separation, NMF is used to decompose the near-end microphone signal onto the union of two nonnegative bases in the magnitude Short Time Fourier Transform domain. One of these bases is for the spectral energy of the acoustic echo signal, and is formed from the in- coming far-end user’s speech, while the other basis is for the spectral energy of the near-end speaker, and is trained with speech data a priori. In comparison to AEC, the speaker extraction approach obviates Double-Talk Detection (DTD), and is demonstrated to attain its maximal echo mitigation performance immediately upon initiation and to maintain that performance during and after room changes for similar computational requirements. Speaker extraction is also shown to introduce distortion of the near-end speech signal during double-talk, which is quantified by means of a speech distortion measure and compared to that of AEC. Subsequently, we address Double-Talk Detection (DTD) for block-based AEC algorithms. We propose a novel block-based DTD algorithm that uses the available signals and the estimate of the echo signal that is produced by NMF-based speaker extraction to compute a suitably normalized correlation-based decision variable, which is compared to a fixed threshold to decide on doubletalk. Using a standard evaluation technique, the proposed algorithm is shown to have comparable detection performance to an existing conventional block-based DTD algorithm. It is also demonstrated to inherit the room change insensitivity of speaker extraction, with the proposed DTD algorithm generating minimal false doubletalk indications upon initiation and in response to room changes in comparison to the existing conventional DTD. We also show that this property allows its paired AEC to converge at a rate close to the optimum. Another focus of this thesis is the problem of inverting a single measurement of a non- minimum phase Room Impulse Response (RIR). We describe the process by which percep- tually detrimental all-pass phase distortion arises in reverberant speech filtered by the inverse of the minimum phase component of the RIR; in short, such distortion arises from inverting the magnitude response of the high-Q maximum phase zeros of the RIR. We then propose two novel partial inversion schemes that precisely mitigate this distortion. One of these schemes employs NMF-based MSSS to separate the all-pass phase distortion from the target speech in the magnitude STFT domain, while the other approach modifies the inverse minimum phase filter such that the magnitude response of the maximum phase zeros of the RIR is not fully compensated. Subjective listening tests reveal that the proposed schemes generally produce better quality output speech than a comparable inversion technique

An investigation of the utility of monaural sound source separation via nonnegative matrix factorization applied to acoustic echo and reverberation mitigation for hands-free telephony

In this thesis we investigate the applicability and utility of Monaural Sound Source Separation (MSSS) via Nonnegative Matrix Factorization (NMF) for various problems related to audio for hands-free telephony. We first investigate MSSS via NMF as an alternative acoustic echo reduction approach to existing approaches such as Acoustic Echo Cancellation (AEC). To this end, we present the single-channel acoustic echo problem as an MSSS problem, in which the objective is to extract the users signal from a mixture also containing acoustic echo and noise. To perform separation, NMF is used to decompose the near-end microphone signal onto the union of two nonnegative bases in the magnitude Short Time Fourier Transform domain. One of these bases is for the spectral energy of the acoustic echo signal, and is formed from the in- coming far-end user’s speech, while the other basis is for the spectral energy of the near-end speaker, and is trained with speech data a priori. In comparison to AEC, the speaker extraction approach obviates Double-Talk Detection (DTD), and is demonstrated to attain its maximal echo mitigation performance immediately upon initiation and to maintain that performance during and after room changes for similar computational requirements. Speaker extraction is also shown to introduce distortion of the near-end speech signal during double-talk, which is quantified by means of a speech distortion measure and compared to that of AEC. Subsequently, we address Double-Talk Detection (DTD) for block-based AEC algorithms. We propose a novel block-based DTD algorithm that uses the available signals and the estimate of the echo signal that is produced by NMF-based speaker extraction to compute a suitably normalized correlation-based decision variable, which is compared to a fixed threshold to decide on doubletalk. Using a standard evaluation technique, the proposed algorithm is shown to have comparable detection performance to an existing conventional block-based DTD algorithm. It is also demonstrated to inherit the room change insensitivity of speaker extraction, with the proposed DTD algorithm generating minimal false doubletalk indications upon initiation and in response to room changes in comparison to the existing conventional DTD. We also show that this property allows its paired AEC to converge at a rate close to the optimum. Another focus of this thesis is the problem of inverting a single measurement of a non- minimum phase Room Impulse Response (RIR). We describe the process by which percep- tually detrimental all-pass phase distortion arises in reverberant speech filtered by the inverse of the minimum phase component of the RIR; in short, such distortion arises from inverting the magnitude response of the high-Q maximum phase zeros of the RIR. We then propose two novel partial inversion schemes that precisely mitigate this distortion. One of these schemes employs NMF-based MSSS to separate the all-pass phase distortion from the target speech in the magnitude STFT domain, while the other approach modifies the inverse minimum phase filter such that the magnitude response of the maximum phase zeros of the RIR is not fully compensated. Subjective listening tests reveal that the proposed schemes generally produce better quality output speech than a comparable inversion technique