Processamento eficiente de arranjos de microfones modulados em densidade de pulso

Orientador: Bruno Sanches MasieroDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de ComputaçãoResumo: Atualmente, os microfones digitais modulados por densidade de pulso (PDM) são amplamente utilizados em aplicações comerciais, já que esta é uma maneira eficiente de transmitir informação de áudio para processadores digitais em dispositivos móveis. No entanto, como o estado-da-arte em algoritmos de processamento digital de arranjos assume que todos os sinais recebidos dos microfones estão em uma representação em banda-base, estes microfones digitais requerem custosos filtros de decimação de alta ordem para converter o fluxo PDM para a modulação por código de pulso (PCM) em banda base. Assim, a implementação destes algoritmos em sistemas embarcados, onde os recursos de processamento são críticos, ou em circuitos integrados (VLSI), onde a energia consumida e área também são críticas, pode se tornar muito dispendiosa devido ao uso de dezenas de filtros de decimação para converter os sinais de PDM para PCM. Essa dissertação explora e propõe métodos eficientes em recursos para a implementação de arranjo de microfones. Com esse intuito, primeiro explora os atuais métodos de design de filtros de decimação e, baseado neles, propõe um algoritmo para fazer o seu design otimizando área e consumo de potência. Também são discutidas as vantagens e desvantagens de se realizar o processamento de arranjo de microfones diretamente nos sinais PDM ao invés dos sinais em PCM. Finalmente propõe um método eficiente para implementação de arranjos de microfones baseado em filtros maximamente planos (MAXFLAT). Como resultado, um novo método para o design de filtros de decimação que optimiza o número de somas por segundo é proposto, assim como demonstra-se que que um filtro espacial implementado no domínio PDM precisa de menos recursos que outras implementação no domínio do tempo. Conclui-se, portanto, que a implementação baseada em filtros MAXFLAT tem um melhor compromiso entre requisitos de armazenamento e poder de computação que o estado-da-arte e os métodos no domínio do PDMAbstract: Nowadays, pulse-density modulated (PDM) digital microphones are widely used on commercial applications as they have become a popular way to deliver audio to digital processors on mobile applications. However, as state-of-the-art array processing algorithms assume that all microphone signals are available in pulse-code modulated (PCM) representation, these digital microphones require costly high-order decimation filters to translate PDM bitstreams to baseband multi-bit PCM signals. In that manner, the implementation of microphone array algorithms in embedded systems, where processing resources are critical, or in very large-scale integration (VLSI) circuits, where power and area are critical, may become very expensive because of the use of the tens of decimation filters required to convert PDM bitstreams into PCM signals. This thesis explores and proposes resource-efficient methods to implement microphone array beamforming. For this purpose, it first reviews the state-of-the-art decimation filter design methods and proposes an algorithm to design decimation filters optimizing area and power consumption. Then it discusses the trade-offs of doing the beamforming calculations at the PDM bitstreams instead of PCM signals and proposes an architecture to implement beamformers without decimation filters. Finally it proposes an efficient approach to implement beamformers based on maximally flat (MAXFLAT) filters. As a result, a new generalized method to design decimation filters optimizing the number of addition per second is proposed, and it is shown that a beamformer implemented in PDM domain requires less resources for its implementation in time domain than other methods. It is concluded that the proposed MAXFLAT-based approach has better storage versus computation efficiency than state-of-the-art and PDM domain implementation approachesMestradoTelecomunicações e TelemáticaMestre em Engenharia Elétric

Digital signal processing and digital-to-analog converters for wide-band transmitters

In this thesis, the implementation methods of digital signal processing and digital-to-analog converters for wide-band transmitters are researched. With digital signal processing, the problems of analog signal processing, such as sensitivity to interference and nonidealities of the semiconductor processes, can be avoided. Also, the programmability can be implemented digitally more easily than by means of analog signal processing. During the past few years, wireless communications has evolved from analog to digital, and signal bandwidths have increased, enabling faster and faster data transmission. The evolution of semiconductor processes, decreasing linewidth and supply voltages, has decreased the size of the electronics and power dissipation, enabling the integration of larger and larger systems on single silicon chips. There is little overall benefit in decreasing linewidths to meet the needs of analog design, since it makes the design process more difficult as the device sizes cannot be scaled according to minimum linewidth and because of the decreasing supply voltage. On the other hand, the challenges of digital signal processing are related to the efficient realization of signal processing algorithms in such a way that the required area and power dissipation does not increase extensively. In this book, the problems related to digital filters, upconversion algorithms and digital-to-analog converters used in digital transmitters are researched. Research results are applied to the implementation of a transmitter for a third-generation WCDMA base-station. In addition, the theory of factors affecting the linearity and performance of digital-to-analog converters is researched, and a digital calibration algorithm for enhancement of the static linearity has been presented. The algorithm has been implemented together with a 16-bit converter; its functionality has been demonstrated with measurements.Tässä väitöskirjassa on tutkittu digitaalisen signaalinkäsittelyn toteuttamista ja digitaalisesta analogiseksi -muuntimia laajakaistaisiin lähettimiin. Digitaalisella signaalinkäsittelyllä voidaan välttää monia analogiseen signaalinkäsittelyyn liittyviä ongelmia, kuten häiriöherkkyyttä ja puolijohdeprosessien epäideaalisuuksien vaikutuksia. Myös ohjelmoitavuus on helpommin toteutettavissa digitaalisesti kuin analogisen signaalinkäsittelyn keinoin. Viime vuosina on langattomien tietoliikennejärjestelmien kehitys kulkenut analogisesta digitaaliseen, ja käytettävät signaalikaistanleveydet ovat kasvaneet mahdollistaen yhä nopeamman tiedonsiirron. Puolijohdeprosessien kehitys, kapeneva minimiviivanleveys ja pienemmät käyttöjännitteet, on pienentänyt elektroniikan kokoa ja tehonkulutusta mahdollistaen yhä suurempien kokonaisuuksien integroimisen yhdelle piisirulle. Viivanleveyksien pieneneminen ei kuitenkaan suoraan hyödytä analogiasuunnittelua, jossa piirielementtien kokoa ei välttämättä voida pienentää viivanleveyden pienentyessä, ja jossa madaltuva käyttöjännite ennemminkin hankaloittaa kuin helpottaa suunnittelua. Siksi yhä suurempi osa signaalinkäsittelystä pyritään tekemään digitaalisesti. Digitaalisen signaalinkäsittelyn ongelmat puolestaan liittyvät algoritmien tehokkaaseen toteuttamiseen siten, että piirien pinta-ala ja tehonkulutus eivät kasva liian suuriksi. Tässä kirjassa on tutkittu digitaalisessa lähettimessä tarvittavien digitaalisten suodattimien, ylössekoitusalgoritmien ja digitaalisesta analogiseksi -muuntimien toteuttamiseen liittyviä ongelmia. Tutkimustuloksia on sovellettu kolmannen sukupolven WCDMA-tukiasemalähettimen toteutuksessa. Lisäksi on tutkittu digitaalisesta analogiseksi -muuntimien lineaarisuuteen ja suorituskykyyn vaikuttavien seikkojen teoriaa, ja esitetty digitaalinen kalibrointialgoritmi muuntimen staattisen suorituskyvyn parantamiseksi. Algoritmi on toteutettu 16-bittisen muuntimen yhteydessä ja se on osoitettu toimivaksi mittauksin.reviewe

Timing Signals and Radio Frequency Distribution Using Ethernet Networks for High Energy Physics Applications

Timing networks are used around the world in various applications from telecommunications systems to industrial processes, and from radio astronomy to high energy physics. Most timing networks are implemented using proprietary technologies at high operation and maintenance costs. This thesis presents a novel timing network capable of distributed timing with subnanosecond accuracy. The network, developed at CERN and codenamed “White- Rabbit”, uses a non-dedicated Ethernet link to distribute timing and data packets without infringing the sub-nanosecond timing accuracy required for high energy physics applications. The first part of this thesis proposes a new digital circuit capable of measuring time differences between two digital clock signals with sub-picosecond time resolution. The proposed digital circuit measures and compensates for the phase variations between the transmitted and received network clocks required to achieve the sub-nanosecond timing accuracy. Circuit design, implementation and performance verification are reported. The second part of this thesis investigates and proposes a new method to distribute radio frequency (RF) signals over Ethernet networks. The main goal of existing distributed RF schemes, such as Radio-Over-Fibre or Digitised Radio-Over-Fibre, is to increase the bandwidth capacity taking advantage of the higher performance of digital optical links. These schemes tend to employ dedicated and costly technologies, deemed unnecessary for applications with lower bandwidth requirements. This work proposes the distribution of RF signals over the “White-Rabbit” network, to convey phase and frequency information from a reference base node to a large numbers of remote nodes, thus achieving high performance and cost reduction of the timing network. Hence, this thesis reports the design and implementation of a new distributed RF system architecture; analysed and tested using a purpose-built simulation environment, with results used to optimise a new bespoke FPGA implementation. The performance is evaluated through phase-noise spectra, the Allan-Variance, and signalto- noise ratio measurements of the distributed signals

Direct digital synthesizers : theory, design and applications

Traditional designs of high bandwidth frequency synthesizers employ the use of a phase-locked-loop (PLL). A direct digital synthesizer (DDS) provides many significant advantages over the PLL approaches. Fast settling time, sub-Hertz frequency resolution, continuous-phase switching response and low phase noise are features easily obtainable in the DDS systems. Although the principle of the DDS has been known for many years, the DDS did not play a dominant role in wideband frequency generation until recent years. Earlier DDSs were limited to produce narrow bands of closely spaced frequencies, due to limitations of digital logic and D/A-converter technologies. Recent advantages in integrated circuit (IC) technologies have brought about remarkable progress in this area. By programming the DDS, adaptive channel bandwidths, modulation formats, frequency hopping and data rates are easily achieved. This is an important step towards a "software-radio" which can be used in various systems. The DDS could be applied in the modulator or demodulator in the communication systems. The applications of DDS are restricted to the modulator in the base station. The aim of this research was to find an optimal front-end for a transmitter by focusing on the circuit implementations of the DDS, but the research also includes the interface to baseband circuitry and system level design aspects of digital communication systems. The theoretical analysis gives an overview of the functioning of DDS, especially with respect to noise and spurs. Different spur reduction techniques are studied in detail. Four ICs, which were the circuit implementations of the DDS, were designed. One programmable logic device implementation of the CORDIC based quadrature amplitude modulation (QAM) modulator was designed with a separate D/A converter IC. For the realization of these designs some new building blocks, e.g. a new tunable error feedback structure and a novel and more cost-effective digital power ramp generator, were developed.reviewe

Algorithms and Circuits for Analog-Digital Hybrid Multibeam Arrays

Fifth generation (5G) and beyond wireless communication systems will rely heavily on larger antenna arrays combined with beamforming to mitigate the high free-space path-loss that prevails in millimeter-wave (mmW) and above frequencies. Sharp beams that can support wide bandwidths are desired both at the transmitter and the receiver to leverage the glut of bandwidth available at these frequency bands. Further, multiple simultaneous sharp beams are imperative for such systems to exploit mmW/sub-THz wireless channels using multiple reflected paths simultaneously. Therefore, multibeam antenna arrays that can support wider bandwidths are a key enabler for 5G and beyond systems. In general, N-beam systems using N-element antenna arrays will involve circuit complexities of the order of N2. This dissertation investigates new analog, digital and hybrid low complexity multibeam beamforming algorithms and circuits for reducing the associated high size, weight, and power (SWaP) complexities in larger multibeam arrays. The research efforts on the digital beamforming aspect propose the use of a new class of discrete Fourier transform (DFT) approximations for multibeam generation to eliminate the need for digital multipliers in the beamforming circuitry. For this, 8-, 16- and 32-beam multiplierless multibeam algorithms have been proposed for uniform linear array applications. A 2.4 GHz 16-element array receiver setup and a 5.8 GHz 32-element array receiver system which use field programmable gate arrays (FPGAs) as digital backend have been built for real-time experimental verification of the digital multiplierless algorithms. The multiplierless algorithms have been experimentally verified by digitally measuring beams. It has been shown that the measured beams from the multiplierless algorithms are in good agreement with the exact counterpart algorithms. Analog realizations of the proposed approximate DFT transforms have also been investigated leading to low-complex, high bandwidth circuits in CMOS. Further, a novel approach for reducing the circuit complexity of analog true-time delay (TTD) N-beam beamforming networks using N-element arrays has been proposed for wideband squint-free operation. A sparse factorization of the N-beam delay Vandermonde beamforming matrix is used to reduce the total amount of TTD elements that are needed for obtaining N number of beams in a wideband array. The method has been verified using measured responses of CMOS all-pass filters (APFs). The wideband squint-free multibeam algorithm is also used to propose a new low-complexity hybrid beamforming architecture targeting future 5G mmW systems. Apart from that, the dissertation also explores multibeam beamforming architectures for uniform circular arrays (UCAs). An algorithm having N log N circuit complexity for simultaneous generation of N-beams in an N-element UCA is explored and verified

Applications in Electronics Pervading Industry, Environment and Society

This book features the manuscripts accepted for the Special Issue “Applications in Electronics Pervading Industry, Environment and Society—Sensing Systems and Pervasive Intelligence” of the MDPI journal Sensors. Most of the papers come from a selection of the best papers of the 2019 edition of the “Applications in Electronics Pervading Industry, Environment and Society” (APPLEPIES) Conference, which was held in November 2019. All these papers have been significantly enhanced with novel experimental results. The papers give an overview of the trends in research and development activities concerning the pervasive application of electronics in industry, the environment, and society. The focus of these papers is on cyber physical systems (CPS), with research proposals for new sensor acquisition and ADC (analog to digital converter) methods, high-speed communication systems, cybersecurity, big data management, and data processing including emerging machine learning techniques. Physical implementation aspects are discussed as well as the trade-off found between functional performance and hardware/system costs

Novel load identification techniques and a steady state self-tuning prototype for switching mode power supplies

Control of Switched Mode Power Supplies (SMPS) has been traditionally achieved through analog means with dedicated integrated circuits (ICs). However, as power systems are becoming increasingly complex, the classical concept of control has gradually evolved into the more general problem of power management, demanding functionalities that are hardly achievable in analog controllers. The high flexibility offered by digital controllers and their capability to implement sophisticated control strategies, together with the programmability of controller parameters, make digital control very attractive as an option for improving the features of dcdc converters. On the other side, digital controllers find their major weak point in the achievable dynamic performances of the closed loop system. Indeed, analogto-digital conversion times, computational delays and sampling-related delays strongly limit the small signal closed loop bandwidth of a digitally controlled SMPS. Quantization effects set other severe constraints not known to analog solutions. For these reasons, intensive scientific research activity is addressing the problem of making digital compensator stronger competitors against their analog counterparts in terms of achievable performances. In a wide range of applications, dcdc converters with high efficiency over the whole range of their load values are required. Integrated digital controllers for Switching Mode Power Supplies are gaining growing interest, since it has been shown the feasibility of digital controller ICs specifically developed for high frequency switching converters. One very interesting potential benefit is the use of autotuning of controller parameters (on-line controllers), so that the dynamic response can be set at the software level, independently of output capacitor filters, component variations and ageing. These kind of algorithms are able to identify the output filter configuration (system identification) and then automatically compute the best compensator gains to adjust system margins and bandwidth. In order to be an interesting solution, however, the self-tuning should satisfy two important requirements: it should not heavily affect converter operation under nominal condition and it should be based on a simple and robust algorithm whose complexity does not require a significant increase of the silicon area of the IC controller. The first issue is avoided performing the system identification (SI) with the system open loop configuration, where perturbations can be induced in the system before the start up. Much more challenging is to satisfy this requirement during steady state operations, where perturbations on the output voltage are limited by the regular operations of the converter. The main advantage of steady state SI methods, is the detection of possible non-idealities occurring during the converter operations. In this way, the system dynamics can be consequently adjusted with the compensator parameters tuning. The resource saving issue, requires the development of äd-hocßelf-tuning techniques specifically tailored for integrated digitally controlled converters. Considering the flexibility of digital control, self-tuning algorithms can be studied and easily integrated at hardware level into closed loop SMPS reducing development time and R & D costs. The work of this dissertation finds its origin in this context. Smart power management is accomplished by tuning the controller parameters accordingly to the identified converter configuration. Themain difficult for self-tuning techniques is the identification of the converter output filter configuration. Two novel system identification techniques have been validated in this dissertation. The open loop SI method is based on the system step response, while dithering amplification effects are exploited for the steady state SI method. The open loop method can be used as autotunig approach during or before the system start up, a step evolving reference voltage has been used as system perturbation and to obtain the output filter information with the Power Spectral Density (PSD) computation of the system step response. The use of ¢§ modulator is largely increasing in digital control feedback. During the steady state, the finite resolution introduces quantization effects on the signal path causing low frequency contributes of the digital control word. Through oversampling-dithering capabilities of ¢§ modulators, resolution improvements are obtained. The presented steady state identification techniques demonstrates that, amplifying the dithering effects on the signal path, the output filter information can be obtained on the digital side by processing with the PSD computation the perturbed output voltage. The amount of noise added on the output voltage does not affect the converter operations, mathematical considerations have been addressed and then justified both with a Matlab/Simulink fixed-point and a FPGA-based closed loop system. The load output filter identification of both algorithms, refer to the frequency domain. When the respective perturbations occurs, the system response is observed on the digital side and processed with the PSD computation. The extracted parameters are the resonant frequency ans the possible ESR (Effective Series Resistance) contributes,which can be detected as maximumin the PSD output. The SI methods have been validated for different configurations of buck converters on a fixed-point closed loop model, however, they can be easily applied to further converter configurations. The steady state method has been successfully integrated into a FPGA-based prototype for digitally controlled buck converters, that integrates a PSD computer needed for the load parameters identification. At this purpose, a novel VHDL-coded full-scalable hybrid processor for Constant Geometry FFT (CG-FFT) computation has been designed and integrated into the PSD computation system. The processor is based on a variation of the conventional algorithm used for FFT, which is the Constant-Geometry FFT (CG-FFT).Hybrid CORDIC-LUT scalable architectures, has been introduced as alternative approach for the twiddle factors (phase factors) computation needed during the FFT algorithms execution. The shared core architecture uses a single phase rotator to satisfy all TF requests. It can achieve improved logic saving by trading off with computational speed. The pipelined architecture is composed of a number of stages equal to the number of PEs and achieves the highest possible throughput, at the expense of more hardware usage