Dissertations / Theses on the topic 'Mixed analogue digital integrate circuit'

La tecnica acustica di verifica sperimentale del range di protoni (ionoacustica) si basa sul rilevamento del debole segnale termoacustico emesso dalla rapida deposizione di energia che avviene alla fine range del fascio, in corrispondenza del picco di Bragg. In questo contesto, questa tesi presenta le principali caratteristiche della strumentazione microelettronica utilizzata per i Proton Sound Detector introducendo specifiche tecniche di progettazione fortemente orientate sia alla massimizzazione del Rapporto Segnale Rumore SNR (a livello di sensore acustico) che minimizzazione della figura di rumore (a livello di amplificatore analogico). La prima parte di questa tesi tratta delle sfide strumentali relative agli esperimenti ionoacustici fornendo dettagli tecnici specifici riguardanti sia la progettazione del sensore acustico (ovvero come costruire il sensore massimizzando l'SNR) sia il design dell'amplificatore a basso rumore (LNA). Verranno presentati i risultati sperimentali di un primo esperimento effettuato presso il Laboratorio Maier-Leibniz di Garching, Monaco, con un fascio di protoni a 20 MeV (scenario preclinico) e verrà mostrato come una progettazione elettronica dedicata a segnali misti permetta di migliorare significativamente il rapporto segnale-rumore e l'accuratezza della localizzazione del picco di Bragg di 6 dB. In questo contesto, questo primo sviluppo del rivelatore raggiunge due importanti obiettivi: il miglioramento dell'SNR a parità di dose e una forte semplificazione della strumentazione del rivelatore rispetto allo stato dell'arte, consentendo una maggiore precisione della misurazione dell'impulso acustico, e allo stesso tempo incrementando la portabilità e la compattezza del dispositivo. Nelle applicazioni cliniche di adroterapia, l'energia del fascio (da 65 MeV fino a 200 MeV) e la dose vengono scelte in funzione dello specifico scenario clinico. Ciò comporta segnali acustici di ampiezza e larghezza di banda diverse, costringendo l’adozione di soluzioni tecnologiche avanzate in grado di gestire un ampio spettro di segnali in termini di larghezza di banda, ampiezza e rumore. Per questo motivo, la seconda parte di questa tesi propone un modello Matlab efficiente e innovativo del fenomeno fisico ionoacustico, che condensa in un unico sistema lineare tempo invariante tutti i processi di conversione dell'energia coinvolti. Il modello ionoacustico proposto sostituisce i complessi strumenti di simulazione classici (usati per caratterizzare il segnale acustico indotto dal fascio di protoni) e facilita lo sviluppo di rivelatori dedicati fornendo una descrizione precisa del segnale acustico nei diversi scenari. Infine, verrà presentato il progetto di una seconda versione del Proton Sound Detector che introduce il concetto di media nel dominio dello spazio (invece della media nel dominio del tempo, basata sull’elaborazione di più shot del fascio che comporta una significativa extra-dose). Questo rilevatore utilizza un sensore multicanale per eseguire una media spaziale dei segnali acquisiti e aumentare l'SNR di 18 dB a parità di dose rispetto al classico approccio monocanale. Questo approccio tuttavia richiede lo sviluppo di elettronica altamente miniaturizzata che non può essere implementata con componenti standard su circuiti stampati. Viene quindi presentato il progetto e la caratterizzazione di un front-end analogico multicanale implementato su un Application-Specified-Integrated-Circuit (ASIC) in tecnologia CMOS 28 nm che permette di elaborare in parallelo tutti i 64 canali del sensore acustico. Questo High-Resolution Proton Sound Detector (HR-ProSD) è completato da un circuito digitale dedicato implementato su FPGA (Field Programmable Gate Array) che consente di mappare in tempo reale e 2D la deposizione di dose nello spazio.
Acoustic proton range experimental verification technique (iono-acoustics) is based on sensing the weak thermoacoustic signal emitted by the fast energy deposition (and/or the heating process) at the end of the beam range (Bragg Peak). In this context, this thesis presents the main characteristics of the micro-electronics instrumentation used for proton sound detectors introducing specific design techniques strongly oriented to both maximization of the acoustic Signal-to-Noise-Ratio (at the Acoustic Sensor level) and Noise-Figure minimization (at analog amplifier level). The first part of this thesis addresses all the instrumentation challenges related to iono-acoustic experiments providing specific technical details regarding both acoustic sensor design (i.e. how to build the sensor while maximizing the SNR) and the LNA design. The experimental results of a first experiment carried out at Maier-Leibniz Laboratory in Garching, Munich, with a proton beam at 20 MeV (sub-clinical energy) will be presented and it will be shown how a dedicated mixed-signal electronics design allows to significantly improve the signal-to-noise ratio and the accuracy of the BP localization by 6 dB. In this context, this first detector development achieves two important objectives: the improvement of the acoustic SNR and a strong simplification of the detector instrumentation w.r.t. state-of-the-art, enabling increasing accuracy of the acoustic pulse measurement, and at the same time the portability and compactness of the device. In clinical hadron-therapy applications, variable beam energy (from 65 MeV up to 200 MeV) and variable doses are used as a function of the selected medical treatment. This induces different acoustic pulses amplitude and bandwidth, forcing advanced technological solutions capable of handling a wide spectrum of signals in terms of bandwidth, amplitude, and noise. For this reason, the second part of this thesis proposes an efficient and innovative Matlab Model of the ionoacoustic physical phenomenon, based on englobing in a single mathematical Linear-Time-Invariant-System all energy conversion processes involved in iono-acoustics. The proposed ionoacoustics model replaces classical and complex simulation tools (used to characterize the proton induced acoustic signal) and facilitates the development of dedicated detectors. Finally, the design of a second version of the Proton Sound Detector will be presented that introduces the concept of space-domain averaging (instead of time-domain averaging based on multiple beam shot processing for noise attenuation and thus extra-doses). This detector uses a multi-channel sensor to perform a spatial average of the acquired signals and increase the SNR by 18 dB at the same dose compared to the classic single channel approach. This approach however requires the development of highly miniaturized electronics that cannot be implemented with off-the-shelf components on Printed Circuit Boards. The design and characterization of a multichannel analog front-end implemented on a CMOS 28 nm Application-Specified-Integrated-Circuit (ASIC) which allows to process the 64 channels of the acoustic sensor in parallel is then presented. This High-Resolution Proton Sound Detector (HR-ProSD) is completed by digital circuits implemented on Field Programmable Gate Array (FPGA) that allow to locate in real time the deposition of energy in space.

Consumers are demanding more and more value for each dollar spent on new electronic equipment. Built-In Self-Test (BIST) of electronic circuits and equipment will help to satiate the demand for self test, self diagnostics, and self repair. This dissertation explores a technique for a Mixed Analog Digital BIST (MADBIST) on a mixed-signal Integrated Circuit (IC). Specifically, on-chip tests for the Analog-to-Digital Converter and Digital-to-Analog Converter on the mixed-signal IC are developed. (The digital portion of the IC can be tested using digital BIST techniques). The tests implemented include Frequency Response, Signal-to-Noise Ratio, Gain Tracking, Inter-Modulation Distortion, and Harmonic Distortion. A precision analog test stimulus is efficiently generated on-chip using digital circuitry. The test stimulus itself is encoded within a Pulse-Density-Modulated bit stream. A narrow-band digital filter is employed to extract the measurement results. Experimental results from a test chip and a prototype circuit board are provided. Some of the engineering and economical trade-offs associated with the design of the tests are considered. The overhead required to implement several types of tests is dealt with. We also explore the relationship between the accuracy achieved by the test and the amount of resources required to implement it.

Silicon carbide (SiC) integrated circuits (ICs) can enable the emergence of robust and reliable systems, including data acquisition and on-site control for extreme environments with high temperature and high radiation such as deep earth drilling, space and aviation, electric and hybrid vehicles, and combustion engines. In particular, SiC ICs provide significant benefit by reducing power dissipation and leakage current at temperatures above 300 °C compared to the Si counterpart. In fact, Si-based ICs have a limited maximum operating temperature which is around 300 °C for silicon on insulator (SOI). Owing to its superior material properties such as wide bandgap, three times larger than Silicon, and low intrinsic carrier concentration, SiC is an excellent candidate for high-temperature applications. In this thesis, analog and mixed-signal circuits have been implemented using SiC bipolar technology, including bandgap references, amplifiers, a master-slave comparator, an 8-bit R-2R ladder-based digital-to-analog converter (DAC), a 4-bit flash analog-to-digital converter (ADC), and a 10-bit successive-approximation-register (SAR) ADC. Spice models were developed at binned temperature points from room temperature to 500 °C, to simulate and predict the circuits’ behavior with temperature variation. The high-temperature performance of the fabricated chips has been investigated and verified over a wide temperature range from 25 °C to 500 °C. A stable gain of 39 dB was measured in the temperature range from 25 °C up to 500 °C for the inverting operational amplifier with ideal closed-loop gain of 40 dB. Although the circuit design in an immature SiC bipolar technology is challenging due to the low current gain of the transistors and lack of complete AC models, various circuit techniques have been applied to mitigate these problems. This thesis details the challenges faced and methods employed for device modeling, integrated circuit design, layout implementation and finally performance verification using on-wafer characterization of the fabricated SiC ICs over a wide temperature range.

QC 20170905

High-speed and high-resolution digital-to-analog converters (DACs) are vital components in all telecommunication systems. Radio-frequency digital-to-analog converter (RFDAC) provides high-speed and high-resolution conversion from digital domain to an analog signal. RFDACs can be employed in direct-conversion radio transmitter architectures. The idea of RFDAC is to utilize an oscillatory pulse-amplitude modulation instead of the conventional zero-order hold pulse amplitude modulation, which results in DAC output spectrum to have high energy high-frequency lobe, other than the Nyquist main lobe. The frequency of the oscillatory pulse can be chosen, with respect to the sample frequency, such that the aliasing images of the signal at integer multiples of the sample frequency are landed in the high-energy high-frequency lobes of the DAC frequency response. Therefore the high-frequency images of the signal can be used as the output of the DAC, i.e., no need to the mixing stage for frequency up-conversion after the DAC in the radio transmitter. The mixing stage however is not eliminated but it is rather moved into the DAC elements and therefore the local oscillator (LO) signal with high frequency should be delivered to each individual DAC element. In direct-conversion architecture of IQ modulators which utilize the RFDAC technique, however, there is a problem of finite image rejection. The origin of this problem is the different polarity of the spectral response of the oscillatory pulse-amplitude modulation in I and Q branches. The conditions where this problem can be alleviated in IQ modulator employing RFDACs is also discussed in this work. ΣΔ modulators are used preceding the DAC in the transmitter chain to reduce the digital signal’s number of bits, still maintain the same resolution. By utilizing the ΣΔ modulator now the total number of DAC elements has decreased and therefore the delivery of the high-frequency LO signal to each DAC element is practical. One of the costs of employing ΣΔ modulator, however, is a higher quantization noise power at the output of the DAC. The quantization noise is ideally spectrally shaped to out-of-band frequencies by the ΣΔ modulator. The shaped noise which usually has comparatively high power must be filtered out to fulfill the radio transmission spectral mask requirement. Semi-digital FIR filter can be used in the context of digital-to-analog conversion, cascaded with ΣΔ modulator to filter the out-of-band noise by the modulator. In the same time it converts the signal from digital domain to an analog quantity. In general case, we can have a multi-bit, semi-digital FIR filter where each tap of the filter is realized with a sub-DAC of M bits. The delay elements are also realized with M-bit shift registers. If the output of the modulator is given by a single bit, the semi-digital FIR filter taps are simply controlled by a single switch assuming a current-steering architecture DAC. One of the major advantages is that the static linearity of the DAC is optimum. Since there are only two output levels available in the DAC, the static transfer function, regardless of the mismatch errors, is always given by a straight line. In this work, the design of SDFIR filter is done through an optimization procedure where the ΣΔ noise transfer function is also taken into account. Different constraints are defined for different applications in formulation of the SDFIR optimization problem. For a given radio transmitter application the objective function can be defined as, e.g., the hardware cost for SDFIR implementation while the constraint can be set to fulfill the radio transmitter spectral emission mask.

Abstract The static linearity testing of analog-to-digital and digital-to-analog converters (ADCs and DACs) has traditionally required test instruments with higher linearity and resolution than that of the device under test. In this thesis ways to test converters without expensive precision instruments are studied. A novel calculation algorithm for the ADC differential non-linearity (DNL) and integral non-linearity (INL) estimation is proposed. The algorithm assumes that two stimuli with constant offset between them are applied to the ADC under test and that the code density histograms for both stimuli are recorded. The probability density function (PDF) of the stimulus is then solved using simple calculations so that DNL and INL of the ADC can be estimated without a priori known stimuli. If a DAC is used to generate the stimulus to ADC, all inputs and outputs are digital and the new algorithm can be used to obtain the PDF of the DAC output. Moreover, the PDF of DAC actually characterizes its INL and DNL so that this all-digital test configuration enables a simultaneous testing of both converters thanks to the new algorithm. The proposed algorithm is analyzed thoroughly both mathematically and by carrying out several simulations and experimental tests. On the basis of the analysis it is possible to approximate the impending estimation error and select the optimal value for the offset between the stimuli. In theory, the accuracy of the algorithm proposed equals that of the standard histogram method with ideal stimulus, but in practice, the accuracy is limited by that of the offset between the stimuli. Therefore, special attention is paid to development of an accurate and small offset generator which enables ratiometric test setup and solves the problems in the case of reference voltage drift. The proposed on-chip offset generator is built using only four resistors and switches. It occupies 122·22 μm2 in a 130 nm CMOS process and accuracy is appropriate for the INL testing of 12-bit converters from rail-to-rail. Based on the analysis of the influence of resistor non-linearity on the accuracy of offset, it is possible to improve the offset generator further. With discrete resistors, the INL of 16-bit ADCs was tested using a 12-bit signal generator. The proposed simple algorithm and tiny offset generator are considered to be important steps towards built-in DNL and INL testing of ADCs and DACs.

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Conselho Nacional de Desenvolvimento Científico e Tecnológico (CNPq)
Nesta tese, propõe-se um sintetizador de freqüência baseado em phase locked loops (PLL) usando uma arquitetura que utiliza um dual-path loop filter, constituído de componentes passivos e um integrador digital. A proposta é empregar técnicas digitais, para reduzir o custo da implementação do sintetizador de freqüência, e flexibilizar o projeto do loop filter, para possibilitar que a arquitetura opere em uma faixa de freqüência larga de operação e com redução de tons espúrios. O loop filter digital é constituído de um contador crescente/ decrescente cujo clock é proveniente da amostragem da diferença de fase de entrada. As técnicas digitais aplicadas ao loop filter se baseiam em alterações da operação do contador, em tempos pré-estabelecidos, os quais são controlados digitalmente. Essas técnicas possibilitam reduzir o tempo de estabelecimento do PLL ao mesmo tempo em que problemas de estabilidade são resolvidos. No desenvolvimento da técnica de dual-path foi realizado o estudo de sua estabilidade, primeiramente, considerando a aproximação do PLL para um sistema linear e depois usando controle digital. Nesse estudo foram deduzidas as equações do sistema, no domínio contínuo e discreto, tanto para o projeto da estabilidade, quanto para descrever o comportamento do PLL. A metodologia top-down é usada no projeto do circuito integrado. As simulações em nível de sistema são usadas, primeiramente, para as criações das técnicas e posteriormente para a verificação do seu comportamento, usando modelos calibrados com os blocos projetados em nível de transistor. O circuito integrado é proposto para ser aplicado em identificação por rádio freqüência (RFID) na banda de UHF (Ultra High Frequency), usando multi-standard, e deve operar na faixa de 850 MHz a 1010 MHz. O sintetizador de freqüência foi projetado na tecnologia CMOS...
In this thesis, a frequency synthesizers phase locked loops (PLL) based with an architecture that uses a dual-path loop filter consisting of passive components and a digital integrator are proposed. The objective is to employ digital techniques to reduce the implementation cost and get loop filter design flexibility to enable the architecture to have a large tuning range operation and spurious reduction. The digital loop filter is based in an up/down counter where the phase difference is sampled to generate the clock of the counter. The techniques applied in the digital path are based in digitally controlled changes in the counter operation in predefined time points. These techniques provide PLL settling time reductions whiling the stability issues are solved. The stability study of the proposed dual path has been developed. First the linear system approximation for the PLL has been assumed and then employing digital control. The continuous and discrete time equations of architecture were derived in that study applied to stability design as well as to describe the architecture behavior. The top-down methodology has been applied to the integrated circuit design. In the beginning, the system level simulations are used for the techniques creation and then the behavioral models that were calibrated with transistor level blocks are simulated. The application of the circuit is proposed to Radio Frequency Identification (RFID) using UHF (Ultra High Frequency) band for multi-standards application and will operate in range of 850 MHz to 1010 MHz. The proposed frequency synthesizer has been designed in the AMS 0.35 μm CMOS technology with 2V power supply. A 300 μs of settling time and 140 Hz of resolution was obtained in simulations. The proposed frequency synthesizer have low complexity and shown a reference noise suppression about 45.6 dB better than the conventional architecture

Orientador: Nobuo Oki
Banca: Suely Cunha Amaro Mantovani
Banca: Jozué Vieira Filho
Banca: Marcelo Arturo Jara Perez
Banca: Paulo Augusto Dal fabbro
Resumo: Nesta tese, propõe-se um sintetizador de freqüência baseado em phase locked loops (PLL) usando uma arquitetura que utiliza um dual-path loop filter, constituído de componentes passivos e um integrador digital. A proposta é empregar técnicas digitais, para reduzir o custo da implementação do sintetizador de freqüência, e flexibilizar o projeto do loop filter, para possibilitar que a arquitetura opere em uma faixa de freqüência larga de operação e com redução de tons espúrios. O loop filter digital é constituído de um contador crescente/ decrescente cujo clock é proveniente da amostragem da diferença de fase de entrada. As técnicas digitais aplicadas ao loop filter se baseiam em alterações da operação do contador, em tempos pré-estabelecidos, os quais são controlados digitalmente. Essas técnicas possibilitam reduzir o tempo de estabelecimento do PLL ao mesmo tempo em que problemas de estabilidade são resolvidos. No desenvolvimento da técnica de dual-path foi realizado o estudo de sua estabilidade, primeiramente, considerando a aproximação do PLL para um sistema linear e depois usando controle digital. Nesse estudo foram deduzidas as equações do sistema, no domínio contínuo e discreto, tanto para o projeto da estabilidade, quanto para descrever o comportamento do PLL. A metodologia top-down é usada no projeto do circuito integrado. As simulações em nível de sistema são usadas, primeiramente, para as criações das técnicas e posteriormente para a verificação do seu comportamento, usando modelos calibrados com os blocos projetados em nível de transistor. O circuito integrado é proposto para ser aplicado em identificação por rádio freqüência (RFID) na banda de UHF (Ultra High Frequency), usando multi-standard, e deve operar na faixa de 850 MHz a 1010 MHz. O sintetizador de freqüência foi projetado na tecnologia CMOS... (Resumo completo, clicar acesso eletrônico abaixo)
Abstract: In this thesis, a frequency synthesizers phase locked loops (PLL) based with an architecture that uses a dual-path loop filter consisting of passive components and a digital integrator are proposed. The objective is to employ digital techniques to reduce the implementation cost and get loop filter design flexibility to enable the architecture to have a large tuning range operation and spurious reduction. The digital loop filter is based in an up/down counter where the phase difference is sampled to generate the clock of the counter. The techniques applied in the digital path are based in digitally controlled changes in the counter operation in predefined time points. These techniques provide PLL settling time reductions whiling the stability issues are solved. The stability study of the proposed dual path has been developed. First the linear system approximation for the PLL has been assumed and then employing digital control. The continuous and discrete time equations of architecture were derived in that study applied to stability design as well as to describe the architecture behavior. The top-down methodology has been applied to the integrated circuit design. In the beginning, the system level simulations are used for the techniques creation and then the behavioral models that were calibrated with transistor level blocks are simulated. The application of the circuit is proposed to Radio Frequency Identification (RFID) using UHF (Ultra High Frequency) band for multi-standards application and will operate in range of 850 MHz to 1010 MHz. The proposed frequency synthesizer has been designed in the AMS 0.35 μm CMOS technology with 2V power supply. A 300 μs of settling time and 140 Hz of resolution was obtained in simulations. The proposed frequency synthesizer have low complexity and shown a reference noise suppression about 45.6 dB better than the conventional architecture
Doutor

Sources of flicker noise generation in the cross-coupled negative resistance oscillator (NMOS, PMOS, and CMOS) are explored. Also, prior and current work in the area of phase noise modeling is reviewed, including the work of Leeson, Hajimiri, Hegazi, and others, seeking the mechanisms by which flicker noise is upconverted. A Figure of Merit (FOM) methodology suitable to the 1/f3 phase noise region is also developed, which allows a new quantity, FOM1, to be defined. FOM1 is proportional to flicker noise upconverted, thus allowing the effectiveness of flicker noise upconversion suppression techniques to be evaluated, despite possibly changing bias points or tank Q, which would change phase noise and FOM in the 1/f2 region. The work of Hajimiri is extended with a simple Amplitude ISF DC component estimator for the special case of LC CMOS oscillators. A method of adaptive control of an oscillator core is presented, as well, comprised of a CMOS oscillator with a digitally adjustable N and P width, and a circuit (which is essentially a tracking ADC) which repeatedly adjusts the relative N to P width dependent on the estimate to maintain the condition of minimum flicker noise upconversion. A fixed calibration constant is sufficient to allow convergence to within 0.7dB of optimal FOM1 for all cases of N width, for a varactorless oscillator test cell. Finally, a circuit is proposed which would allow the flicker noise reduction technique of cycling to accumulation to be applied to continuous time oscillators, but is not rigorously vetted.

In order to improve the sensitivity of an optical receiver, the gain and the collection area of the photo-detectors within the receiver should be increased. Detectors with internal gain such as avalanche photodiodes (APD) are usually used to increase the sensitivity of the receiver. One problem with APDs is the sensitivity of their gain to their bias voltage, which makes them challenging to be fabricated in a standard CMOS process due to variations in their gain. However, when an APD is biased over its breakdown voltage, it is sensitive to a single photon, hence, referred to as a single photon avalanche diodes (SPAD). The SPADs are photon-counting detectors, which are less sensitive to their bias voltage, and can be integrated with rest of the electronic circuitry that form an optical receiver. An avalanche diode requires dedicated circuits to be operated in the SPAD mode. These circuits make the diode insensitive to an incident photon for a duration that is known as deadtime. Unfortunately, The collection area of the PD, APD, and SPADs are limited to their capacitance. Hence, a large photo-detector leads to a larger capacitance, which reduces the bandwidth of the receiver. In this thesis, a photon counting optical receiver based on an array of SPADs is proposed which increases the collection area with a low output capacitance. The avalanche diode and peripheral circuits which operate and readout-out the SPAD array are fabricated in the commercially available UMC 0.18 μm CMOS process. Initially, the avalanche diode is tested and characterised. A high performance circuit is then designed and tested which is able to achieve short deadtimes up to 4 ns. Once the photon counting operation of the SPAD is verified, a numerical model is developed to investigate the influence of several factors, including the deadtime, on the performance of the photon-counting detector in a communication link. Based on the simulation results, which show the advantages of an array over a single detector, a prototype detector array of 64 asynchronous SPADs is designed and tested. This array uses a high-speed readout mechanism which is inspired by the current steering digital-to-analogue converters. Bit error ratio tests (BERT) verify the photon counting capability of the proposed detector, and a bit error rate of 1E-3 has been achieved at data rate of 100 Mbps. In addition, the array of SPAD is compatible with a front-end of conventional optical receiver which uses a photodiode as a photo detector.

Thesis (M.Eng.(Microelectronic Engineering))--University of Pretoria, 2009.
Includes summaries in Afrikaans and English. Includes bibliographical references (leaves [74]-78). Mode of access: World Wide Web.

High-speed digital to analogue converters (DAC), which are optimised for large bandwidth signal synthesis applications, are a fundamental building block and enabling technology in industrial instrumentation, military, communication and medical applications. The spurious free dynamic range (SFDR) is a key specification of high-speed DACs, as unwanted spurious signals generated by the DAC degrades the performance and effectiveness of wideband systems. The focus of this work is to enhance the SFDR performance of high-speed DACs. As bandwidth requirements increase, meeting the desired SFDR performance is further complicated by the increase in dynamic non-linearity. The most widely used architecture in high-speed applications is the current-steering DAC fabricated on CMOS technology. The current source finite output impedance, switch distortion and clock feedthrough are the greatest contributors to dynamic non-linearity and are difficult to improve with the use of MOS devices alone. This research proposes the use of BiCMOS technology that offers high performance, using heterojunction bipolar transistors (HBT) that, when combined with MOS devices, are able to improve on the linearity of the current-steering DAC and hence improve the SFDR. A design methodology is introduced based on BiCMOS fabrication technology to improve SFDR performance and places emphasis on the constraints of modern fabrication processes. A six-bit current-steering application-specific integrated circuit DAC is designed based on the proposed design methodology, which optimises the SFDR performance of high-speed binary weighted architectures by lowering current switch distortion and reducing the clock feedthrough effect to verify the hypothesis experimentally. A novel current source cell is implemented that comprises HBT current switches, negative channel metal-oxide semiconductor (NMOS) cascode and NMOS current source to overcome distortion by specifically enhancing the SFDR for high-speed DACs. A switch driver and low-voltage differential signalling receiver to achieve high-speed DAC performance and their influence on the SFDR performance are designed and discussed. The DAC is implemented using the International Business Machines Corporation (IBM) 8HP silicon germanium (SiGe) BiCMOS 130 nm technology. The DAC achieves a better than 21.96 dBc SFDR across the Nyquist band for a sampling rate of 500 MS/s with a core size of 0.1 mm2 and dissipates just 4 mW compared to other BiCMOS DACs that achieve similar SFDR performance with higher output voltages, resulting in much larger power dissipation.
Dissertation (MEng)--University of Pretoria, 2015.
Electrical, Electronic and Computer Engineering
MEng
Unrestricted

abstract: Time-interleaved analog to digital converters (ADCs) have become critical components in high-speed communication systems. Consumers demands for smaller size, more bandwidth and more features from their communication systems have driven the market to use modern complementary metal-oxide-semiconductor (CMOS) technologies with shorter channel-length transistors and hence a more compact design. Downscaling the supply voltage which is required in submicron technologies benefits digital circuits in terms of power and area. Designing accurate analog circuits, however becomes more challenging due to the less headroom. One way to overcome this problem is to use calibration to compensate for the loss of accuracy in analog circuits. Time-interleaving increases the effective data conversion rate in ADCs while keeping the circuit requirements the same. However, this technique needs special considerations as other design issues associated with using parallel identical channels emerge. The first and the most important is the practical issue of timing mismatch between channels, also called sample-time error, which can directly affect the performance of the ADC. Many techniques have been developed to tackle this issue both in analog and digital domains. Most of these techniques have high complexities especially when the number of channels exceeds 2 and some of them are only valid when input signal is a single tone sinusoidal which limits the application. This dissertation proposes a sample-time error calibration technique which bests the previous techniques in terms of simplicity, and also could be used with arbitrary input signals. A 12-bit 650 MSPS pipeline ADC with 1.5 GHz analog bandwidth for digital beam forming systems is designed in IBM 8HP BiCMOS 130 nm technology. A front-end sample-and-hold amplifier (SHA) was also designed to compare with an SHA-less design in terms of performance, power and area. Simulation results show that the proposed technique is able to improve the SNDR by 20 dB for a mismatch of 50% of the sampling period and up to 29 dB at 37% of the Nyquist frequency. The designed ADC consumes 122 mW in each channel and the clock generation circuit consumes 142 mW. The ADC achieves 68.4 dB SNDR for an input of 61 MHz.
Dissertation/Thesis
Doctoral Dissertation Electrical Engineering 2017

Includes bibliographical references (leaves 277-284) Presents and discusses new design techniques for mixed analog-digital circuits with emphases on low power and small area for standard low-cost CMOS VLSI technology.

The conventional approach to validate the analog and mixed signal designs utilizes extensive SPICE-level simulations. The main challenge in this approach is to know when all important corner cases have been simulated. An alternate approach is to use the formal verification techniques. Formal verification techniques have gained wide spread popularity in the digital design domain; but in case of analog and mixed signal designs, a large number of test scenarios need to be designed to generate sufficient simulation traces to test out all the specified system behaviours. Analog and mixed signal designs can be formally modeled as hybrid systems and therefore techniques used for formal analysis and verification of hybrid systems can be applied to the analog and mixed signal designs. Generally, formal verification tools for hybrid systems work at the abstract level where we model the systems in terms of differential equations or algebraic equations. However the analog and mixed signal system designers are very comfortable in designing the circuits at the transistor level. To bridge the gap between abstraction level verification and the designs validation which has been implemented at the transistor level, the very important issue we need to address is: Can we formally verify the circuits at the transistor level itself? For this we have proposed a framework for doing the formal verification of analog and mixed signal designs using SPICE simulation traces in one of the hybrid systems formal verification tools (i.e. Checkmate from CMU). An extension to a formal verification approach of hybrid systems is proposed to verify analog and mixed signal (AMS) designs. AMS designs can be formally modeled as hybrid systems and therefore lend themselves to the formal analysis and verification techniques applied to hybrid systems. The proposed approach employs simulation traces obtained from an actual design implementation of AMS circuit blocks (for example, in the form of SPICE netlists) to carry out formal analysis and verification. This enables the same platform used for formally validating an abstract model of an AMS design to be also used for validating its different refinements and design implementation, thereby providing a simple route to formal verification at different levels of implementation. Our approach has been illustrated through the case studies using simulation traces form the different frameworks i.e. Simulink/Stateflow framework and the SPICE simulation traces. We demonstrate the feasibility of our approach around the Checkmate and the case studies for hybrid systems and the analog and mixed signal designs.

Desire for low-power, high performance computing has been at core of the symbiotic union between digital circuits and CMOS scaling. While digital circuit performance improves with device scaling, analog circuits have not gained these benefits. As a result, it has become necessary to leverage increased digital circuit performance to mitigate analog circuit deficiencies in nanometer scale CMOS in order to realize world class analog solutions. In this thesis, both circuit and system enhancement techniques to improve performance of clock generators are discussed. The following techniques were developed: (1) A digital PLL that employs an adaptive and highly efficient way to cancel the effect of supply noise, (2) a supply regulated DPLL that uses low power regulator and improves supply noise rejection, (3) a digital multiplying DLL that obviates the need for high-resolution TDC while achieving sub-picosecond jitter and excellent supply noise immunity, and (4) a high resolution TDC based on a switched ring oscillator, are presented. Measured results obtained from the prototype chips are presented to illustrate the proposed design techniques.
Graduation date: 2013
Access restricted to the OSU Community at author's request from July 16, 2012 - July 16, 2014

The miniaturization of dc–dc switch-mode power supplies (SMPS) is of a key importance in volume-sensitive portable devices, such as cell phones, tablet computers, and digital cameras. In these systems, multiple SMPS are required to provide well regulated voltage and power to various electronic components such as the central processing unit (CPU) and random-access memory (RAM). The combined volume, weight, and surface area footprint of these SMPS is usually the largest component. Traditionally, SMPS volume reduction has been achieved through increased switching frequencies; however, for power-sensitive applications this is undesirable due to the increased switching losses. This thesis presents two alternative, power-efficient, SMPS miniaturization methods: one control and one topology based. The presented controller recovers from load transients with virtually minimum possible output voltage deviation, reducing the reactive component size. The controller utilizes a simple algorithm, requiring no knowledge of the converter parameters and virtually no processing power. The simplicity of the control concept enabled the design of an area and power efficient integrated circuit (IC) implementation. The entire IC is implemented in a CMOS 0.18µm process on a 0.26 mm2 silicon area, which is comparable to the state-of-the-art analog solutions. For the experimental system the deviation (output capacitor size) is about four times smaller than that of a fast PID compensator having a 1/10th of the switching frequency bandwidth. The second solution is a complementary converter topology that has a smaller output filter volume, improved dynamic response, and lower switching losses compared to the state-of-the-art solutions. To reduce the volume and switching losses, the input-to-output voltage difference is decreased with a capacitive attenuator that replaces the input filter capacitor and has approximately the same volume. Both the attenuator and the downstream buck converter share the same set of switches, minimizing conduction losses. A single multi-mode digital controller governs operation of both stages, seamlessly regulating the output and input center-tap voltages. Experiments with a 5–1.5-V, 2.5-A, 1-MHz prototype show that, compared to the conventional buck, the merged topology has 43% smaller inductor, 36% smaller output capacitor, up to 30% lower power losses, and a 25% faster transient response.

Die vorliegende Arbeit behandelt den Entwurf eines ADCs nach dem sukzessiven Approximationsverfahren (SAR). Ausgehend von den Systemanforderungen erfolgt eine Ableitung der Spezifikation des zu entwerfenden ADCs. Theoretische Betrachtungen und Highlevelsimulationen in Matlab wählen die optimale Architektur der Einzelkomponenten - kapazitives DAC Array, Komparator, Ablaufsteuerung - aus. Die Implementation selbst findet für die Analogschaltungsteile auf Transistorebene und für die digitalen Komponenten auf RT-Ebene in VHDL statt. Sie bilden die Grundlage für die Realisierung des Layouts. In dem Zusammenhang stellt die Arbeit die gängigsten Matchingmethoden für elektronische Bauelemente vor. Abschließende PEX-Simulationen (parasitic Extraction) ermitteln die statischen (INL/DNL) wie dynamischen Kennwerte (SNR) des SAR-ADCs.:Abkürzungsverzeichnis iii Formelzeichen v 1 Einleitung 1 2 Grundlagen 2 2.1 Analog/Digital-Umsetzer 2 2.1.1 Umsetzungsverfahren 2 2.1.2 Statische Kennwerte 8 2.1.3 Dynamische Kennwerte 12 2.2 Technologie 17 2.2.1 Übersicht 17 2.2.2 MOS-Transistoren 17 2.2.3 Kapazitäten 18 2.2.4 Widerstände 18 2.3 Hardwarebeschreibungssprache 19 2.3.1 Übersicht 19 2.3.2 Zustandsautomat 19 2.3.3 Look-Ahead-Ausgang 20 3 Spezifikation 21 4 ADU-Topologie 23 4.1 Vorüberlegungen 23 4.1.1 Umsetzungsverfahren 23 4.1.2 Vergleich Widerstand/Kapazität 23 4.1.3 Differenziell Vs. Single-Ended 24 4.1.4 Kapazitätsarray 25 4.2 ADC High-Level Modell 30 4.2.1 Funktionsblöcke 30 4.2.2 Matlab/Simulink 31 4.2.3 Simulation 34 4.3 Parasitäre Effekte 37 4.3.1 Substratkapazität 37 4.3.2 Komparatoroffset 39 5 Schaltungsdesign & -simulation 41 5.1 Komparator 41 5.1.1 Spezifikation 41 5.1.2 Latch 41 5.1.3 Vorverstärker 43 5.1.4 Gesamtsystem 46 5.2 Schalter 46 5.2.1 Funktionsweise 46 5.2.2 Ladungseintrag 46 5.2.3 Dimensionierung & Simulation 47 5.3 Kapazitätsarray 51 5.4 SAR-Controller 51 5.4.1 Vorüberlegung 51 5.4.2 RTL Design 52 5.4.3 Simulation 55 5.4.4 Synthese 57 5.4.5 Optimierung 59 5.5 ADC (Toplevel) 59 5.5.1 Architektur 59 5.5.2 Simulation 61 6 Layout 64 6.1 Komparator 65 6.1.1 Vorverstärker 1 65 6.1.2 Vorverstärker 2 66 6.1.3 Dynamisches Latch 66 6.2 Transmission Gates 67 6.3 Kapazitätsarray 68 6.4 SAR-Controller 70 6.5 ADC (Toplevel) 70 6.6 PEX Simulation 72 6.6.1 Statischer Test 72 6.6.2 Dynamischer Test 73 7 Zusammenfassung 74 Literaturverzeichnis 76 Bücher 76 Skripte und Schriften 76 Internetlinks 78 Abbildungsverzeichnis 79 Tabellenverzeichnis 82 Anhang 84