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  • About
  • The Global ETD Search service is a free service for researchers to find electronic theses and dissertations. This service is provided by the Networked Digital Library of Theses and Dissertations.
    Our metadata is collected from universities around the world. If you manage a university/consortium/country archive and want to be added, details can be found on the NDLTD website.
211

Camera-microcomputer interface

Graham, Helen Louise January 1980 (has links)
Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1980. / MICROFICHE COPY AVAILABLE IN ARCHIVES AND ENGINEERING. / Includes bibliographcial references. / by Helen Louise Graham. / M.S.
212

An IF-input quadrature continuous-time multi-bit [delta][sigma] modulator with high image and non-linearity suppression for dual-standard wireless receiver application.

January 2008 (has links)
Ko, Chi Tung. / On t.p. "delta" and "sigma" appear as the Greek letters. / Thesis submitted in: December 2007. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2008. / Includes bibliographical references. / Abstracts in English and Chinese. / Abstract --- p.1 / 摘要 --- p.3 / Acknowledgements --- p.4 / Table of Contents --- p.5 / List of Figures --- p.8 / List of Tables --- p.13 / Chapter Chapter 1 --- Introduction --- p.14 / Chapter 1.1 --- Motivation --- p.14 / Chapter 1.2 --- Objectives --- p.17 / Chapter 1.3 --- Organization of the Thesis --- p.17 / References --- p.18 / Chapter Chapter 2 --- Fundamentals of Delta-sigma Modulators --- p.20 / Chapter 2.1 --- Delta-sigma Modulator as a Feedback System --- p.20 / Chapter 2.2 --- Quantization Noise --- p.22 / Chapter 2.3 --- Oversampling --- p.23 / Chapter 2.4 --- Noise Shaping --- p.25 / Chapter 2.5 --- Performance Parameters --- p.27 / Chapter 2.6 --- Baseband Modulators vs Bandpass Modulators --- p.27 / Chapter 2.7 --- Discrete-time Modulators vs Continuous-time Modulators --- p.28 / Chapter 2.8 --- Single-bit Modulators vs Multi-bit Modulators --- p.29 / Chapter 2.9 --- Non-linearity and Image Problems in Multi-bit Delta-sigma Modulators --- p.29 / Chapter 2.9.1 --- Non-linearity Problem --- p.29 / Chapter 2.9.2 --- Image Problem --- p.31 / Reference --- p.36 / Chapter Chapter 3 --- Image Rejection and Non-linearity Suppression Techniques for Quadrature Multi-bit Δ¡♭ Modulators --- p.38 / Chapter 3.1 --- Quadrature DEM Technique --- p.38 / Chapter 3.1.1 --- Introduction and Working Principle --- p.38 / Chapter 3.1.2 --- Behavioral Simulation Results --- p.42 / Chapter 3.2 --- IQ DWA Technique --- p.44 / Chapter 3.2.1 --- Introduction and Working Principle --- p.44 / Chapter 3.2.2 --- Behavioral Simulation Results --- p.49 / Chapter 3.3 --- DWA and Bit-wise Data-Dependent DEM --- p.52 / Chapter 3.3.1 --- Introduction and Working Principle --- p.52 / Chapter 3.3.2 --- Behavioral Simulation Results --- p.54 / Chapter 3.4 --- Image Rejection Technique for Quadrature Mixer --- p.61 / Chapter 3.5 --- Conclusion --- p.63 / Reference --- p.64 / Chapter Chapter 4 --- System Design of a Multi-Bit CT Modulator for GSM/WCDMA Application --- p.65 / Chapter 4.1 --- Objective of Design and Design Specification --- p.65 / Chapter 4.2 --- Topology Selection --- p.65 / Chapter 4.3 --- Discrete-time Noise Transfer Function Generation --- p.66 / Chapter 4.4 --- Continuous-time Loop Filter Transfer Function Generation --- p.69 / Chapter 4.5 --- Behavioral Model of Modulator --- p.69 / Chapter 4.6 --- Dynamic Range Scaling --- p.75 / Chapter 4.7 --- Behavioral Modeling of Operational Amplifiers --- p.77 / Chapter 4.8 --- Impact of RC Variation on Performance --- p.85 / Chapter 4.9 --- Loop Filter Component Values --- p.88 / Chapter 4.10 --- Summary --- p.90 / Reference --- p.90 / Chapter Chapter 5 --- Transistor-level Implementation of Modulators --- p.92 / Chapter 5.1 --- Overview of Design --- p.92 / Chapter 5.2 --- Design of Operational Transconductance Amplifiers (OTAs) --- p.94 / Chapter 5.2.1 --- First Stage --- p.94 / Chapter 5.2.2 --- Second and Third Stages --- p.98 / Chapter 5.3 --- Design of Feed-forward Transconductance (Gm) Cells --- p.101 / Chapter 5.4 --- Design of Quantizer --- p.102 / Chapter 5.4.1 --- Reference Ladder Design --- p.102 / Chapter 5.4.2 --- Comparator Design --- p.104 / Chapter 5.5 --- Design of Feedback Digital-to-Analog Converter (DAC) --- p.106 / Chapter 5.5.1 --- DWA and DEM Logic --- p.107 / Chapter 5.5.2 --- DAC Circuit --- p.109 / Chapter 5.6 --- Design of Integrated Mixers --- p.111 / Chapter 5.7 --- Design of Clock Generators --- p.112 / Chapter 5.7.1 --- Master Clock Generator --- p.112 / Chapter 5.7.2 --- LO Clock Generator --- p.114 / Chapter 5.7.3 --- Simulation Results --- p.116 / Reference --- p.125 / Chapter Chapter 6 --- Physical Design of Modulators --- p.127 / Chapter 6.1 --- Floor Planning of Modulator --- p.127 / Chapter 6.2 --- Shielding of Sensitive Signals --- p.130 / Chapter 6.3 --- Common Centroid Layout --- p.130 / Chapter 6.4 --- Amplifier Layout --- p.132 / Reference --- p.137 / Chapter Chapter 7 --- Conclusions --- p.138 / Chapter 7.1 --- Conclusions --- p.138 / Chapter 7.2 --- Future Works --- p.138 / Appendix A Schematics of Building Blocks --- p.140 / First Stage Operational Amplifier --- p.140 / First Stage Amplifier Local Bias Circuit --- p.140 / Second and Third Stage Operational Amplifier --- p.141 / Second and Third Stage Local Bias Circuit --- p.141 / CMFB Circuit (First Stage) --- p.142 / CMFB Circuit (Second Stage) --- p.142 / Gm-Feed-forward Cells --- p.143 / Gm Feed-forward Cell Bias Circuit --- p.143 / Reference Ladder Circuit --- p.144 / Pre-amplifier Circuit --- p.145 / Latch Circuit --- p.145 / DAC Circuit (Unit Cell) --- p.146 / Author's Publications --- p.147
213

Designing and Simulating a Multistage Sampling Rate Conversion System Using a Set of PC Programs

Hagerty, David Joseph 07 May 1993 (has links)
The thesis covers a series of PC programs that we have written that will enable users to easily design FIR linear phase lowpass digital filters and multistage sampling rate conversion systems. The first program is a rewrite of the McClellanParks computer program with some slight modifications. The second program uses an algorithm proposed by Rabiner that determines the length of a lowpass digital filter. Rabiner used a formula proposed by Herrmann et al. to initially estimate the filter length in his algorithm. The formula, however, assumes unity gain. We present a modification to the formula so that the gain of the filter is normalized to accommodate filters that have a gain greater than one (as in the case of a lowpass filter used in an interpolator). We have also changed the input specifications from digital to analog. Thus, the user supplies the sampling rate, passband frequency, stopband frequency, gain, and the respective maximum band errors. The program converts the specifications to digital. Then, the program iteratively estimates the filter length and interacts with the McClellan-Parks Program to determine the actual filter length that minimizes the maximum band errors. Once the actual length is known, the filter is designed and the filter coefficients may be saved to a file. Another new finding that we present is the condition that determines when to add a lowpass filter to a multistage decimator in order to reduce the total number of filter taps required to implement the system. In a typical example, we achieved a 34% reduction in the total required number of filter taps. The third program is a new program that optimizes the design of a multistage sampling rate conversion system based upon the sum of weighted computational rates and storage requirements. It determines the optimum number of stages and the corresponding upsampling and downsampling factors of each stage of the design. It also determines the length of the required lowpass digital filters using the second program. Quantization of the filter coefficients may have a significant impact on the frequency response. Consequently, we have included a routine within our program that determines the effects of such quantization on the allowable error margins within the passband and stopband. Once the filter coefficients are calculated, they can be saved to files and used in an appropriate implementation. The only requirements of the user are the initial sampling rate, final sampling rate, passband frequency, stopband frequency, corresponding maximum errors for each band, and the weighting factors to determine the optimization factor. We also present another new program that implements a sampling rate conversion from CD (44.1 kHz) to DAT (48 kHz) for digital audio. Using the third program to design the filter coefficients, the fourth program converts an input sequence (either samples of a sine wave or a unit sample sequence) sampled at the lower rate to an output sequence sampled at the higher rate. The frequency response is then plotted and the output block may be saved to a file.
214

Low voltage switched capacitor circuits for lowpass and bandpass [delta sigma] converters

Keskin, Mustafa 07 December 2001 (has links)
The most accurate method for performing analog signal processing in MOS (metal-oxide-semiconductor) integrated circuits is through the use of switched-capacitor circuits. A switched-capacitor circuit operates as a discrete-time signal processor. These circuits have been used in a variety of applications, such as filters, gain stages, voltage-controlled oscillators, and modulators. A switched-capacitor circuit contains operational amplifiers (opamps), capacitators, switches, and a clock generator. Capacitors are used to define the state variables of a system. They store charges for a defined time interval, and determine the state variables as voltage differences. Switches are used to direct the flow of charges and to enable the charging and discharging of capacitors. Nonoverlapping clock signals control the switches and allow charge transfer between the capacitors. Opamps are used in order to perform high-accuracy charge transfer from one capacitor to another. The goal of this research is to design and explore future low-voltage switched-capacitor circuits, which are crucial for portable devices. Low-voltage operation is needed for two reasons: making reliable and accurate systems compatible with the submicron CMOS technology and reducing power consumption of the digital circuits. To this end, three different switched-capacitor integrators are proposed, which function with very low supply voltages. One of these configurations is used to design a lowpass ����� modulator for digital-audio applications. This modulator is fabricated and tested demonstrating 80 dB dynamic range with a 1-V supply voltage. The second part of this research is to show that these low-voltage circuits are suitable for modern wireless communication applications, where the clock and signal frequencies are very high. This part of the research has focused on bandpass analog-to-digital converters. Bandpass analog-to-digital converters are among the key components in wireless communication systems. They are used to digitize the received analog signal at an intermediate center frequency. Such converters are used for digital FM or AM radio applications and for portable communication devices, such as cellular phones. The main block, in these converters, is the resonator, which is tuned to a particular center frequency. A resonator must be designed such that it has a sharp peak at a specific center frequency. However, because of circuit imperfections, the resonant peak gain and/or the center frequency are degraded in existing architectures. Two novel switched-capacitor resonators were invented during the second part of this research. These resonators demonstrate superior performance as compared to previous architectures. A fourth-order low-voltage bandpass ����� modulator, using one of these resonators, has been designed. / Graduation date: 2002
215

Non-uniform sampling: algorithms and architectures

Luo, Chenchi 09 November 2012 (has links)
Modern signal processing applications emerging in telecommunication and instrumentation industries have placed an increasing demand for ADCs with higher speed and resolution. The most fundamental challenge in such a progress lies at the heart of the classic signal processing: the Shannon-Nyquist sampling theorem which stated that when sampled uniformly, there is no way to increase the upper frequency in the signal spectrum and still unambiguously represent the signal except by raising the sampling rate. This thesis is dedicated to the exploration of the ways to break through the Shannon-Nyquist sampling rate by applying non-uniform sampling techniques. Time interleaving is probably the most intuitive way to parallel the uniform sampling process in order to achieve a higher sampling rate. Unfortunately, the channel mismatches in the TIADC system make the system an instance of a recurrent non-uniform sampling system whose non-uniformities are detrimental to the performance of the system and need to be calibrated. Accordingly, this thesis proposed a flexible and efficient architecture to compensate for the channel mismatches in the TIADC system. As a key building block in the calibration architecture, the design of the Farrow structured adjustable fractional delay filter has been investigated in detail. A new modified Farrow structure is proposed to design the adjustable FD filters that are optimized for a given range of bandwidth and fractional delays. The application of the Farrow structure is not limited to the design of adjustable fractional delay filters. It can also be used to implement adjustable lowpass, highpass and bandpass filters as well as adjustable multirate filters. This thesis further extends the Farrow structure to the design of filters with adjustable polynomial phase responses. Inspired by the theory of compressive sensing, another contribution of this thesis is to use randomization as a means to overcome the limit of the Nyquist rate. This thesis investigates the impact of random sampling intervals or jitters on the power spectrum of the sampled signal. It shows that the aliases of the original signal can be well shaped by choosing an appropriate probability distribution of the sampling intervals or jitters such that aliases can be viewed as a source of noise in the signal power spectrum. A new theoretical framework has been established to associate the probability mass function of the random sampling intervals or jitters with the aliasing shaping effect. Based on the theoretical framework, this thesis proposes three random sampling architectures, i.e., SAR ADC, ramp ADC and level crossing ADC, that can be easily implemented based on the corresponding standard ADC architectures. Detailed models and simulations are established to verify the effectiveness of the proposed architectures. A new reconstruction algorithm called the successive sine matching pursuit has also been proposed to recover a class of spectrally sparse signals from a sparse set of non-uniform samples onto a denser uniform time grid so that classic signal processing techniques can be applied afterwards.
216

Improving a sampled-data circuit simulator for Delta-Sigma modulator design

Hayward, Roger D. 30 April 1992 (has links)
Delta-Sigma Modulator-based Analog-to-Digital converter design is an active area of research. New topologies require extensive simulations to verify their performance. A series of improvements were made to an existing circuit simulation package in order to speed the simulation process for the designer. Various examples of these improvements are presented in typical applications. / Graduation date: 1992
217

Design of analog-to-digital converters with binary search algorithm and digital calibration techniques

Wong, Si Seng January 2011 (has links)
University of Macau / Faculty of Science and Technology / Department of Electrical and Electronics Engineering
218

Built-in self test of RF subsystems

Zhang, Chaoming, 1980- 04 November 2013 (has links)
With the rapid development of wireless and wireline communications, a variety of new standards and applications are emerging in the marketplace. In order to achieve higher levels of integration, RF circuits are frequently embedded into System on Chip (SoC) or System in Package (SiP) products. These developments, however, lead to new challenges in manufacturing test time and cost. Use of traditional RF test techniques requires expensive high frequency test instruments and long test time, which makes test one of the bottlenecks for reducing IC costs. This research is in the area of built-in self test technique for RF subsystems. In the test approach followed in this research, on-chip detectors are used to calculate circuits specifications, and data converters are used to collect the data for analysis by an on-chip processor. A novel on-chip amplitude detector has been designed and optimized for RF circuit specification test. By using on-chip detectors, both the system performance and specifications of the individual components can be accurately measured. On-chip measurement results need to be collected by Analog to Digital Converters (ADCs). A novel time domain, low power ADC has been designed for this purpose. The ADC architecture is based on a linear voltage controlled delay line. Using this structure results in a linear transfer function for the input dependent delay. The time delay difference is then compared to a reference to generate a digital code. Two prototype test chips were fabricated in commercial CMOS processes. One is for the RF transceiver front end with on-chip detectors; the other is for the test ADC. The 940MHz RF transceiver front-end was implemented with on-chip detectors in a 0.18 [micrometer] CMOS technology. The chips were mounted onto RF Printed Circuit Boards (PCBs), with tunable power supply and biasing knobs. The detector was characterized with measurements which show that the detector keeps linear performance over a wide input amplitude range of 500mV. Preliminary simulation and measurements show accurate transceiver performance prediction under process variations. A 300MS/s 6 bit ADC was designed using the novel time domain architecture in a 0.13 [micrometer] standard digital CMOS process. The simulation results show 36.6dB Signal to Noise Ratio (SNR), 34.1dB Signal to Noise and Distortion Ratio (SNDR) for 99MHz input, Differential Non-Linearity (DNL)<0.2 Least Significant Bit (LSB), and Integral Non-Linearity (INL)<0.5LSB. Overall chip power is 2.7mW with a 1.2V power supply. The built-in detector RF test was extended to a full transceiver RF front end test with a loop-back setup, so that measurements can be made to verify the benefits of the technique. The application of the approach to testing gain, linearity and noise figure was investigated. New detector types are also evaluated. In addition, the low-power delay-line based ADC was characterized and improved to facilitate gathering of data from the detector. Several improved ADC structures at the system level are also analyzed. The built-in detector based RF test technique enables the cost-efficient test for SoCs. / text
219

Discrete-time crossing-point estimation for switching power converters

Smecher, Graeme. January 2008 (has links)
In a number of electrical engineering problems, so-called "crossing points" -- the instants at which two continuous-time signals cross each other -- are of interest. Often, particularly in applications using a Digital Signal Processor (DSP), only periodic samples along with a partial statistical characterization of the signals are available. In this situation, we are faced with the following problem: Given limited information about these signals, how can we efficiently and accurately estimate their crossing points? / For example, an audio amplifier typically receives its input from a digital source decoded into regular samples (e.g. from MP3, DVD, or CD audio), or obtained from a continuous-time signal using an analog-to-digital converter (ADC). In a switching amplifier based on Pulse-Width Modulation (PWM) or Click Modulation (CM), a signal derived from the sampled audio is compared against a deterministic reference waveform; the crossing points of these signals control a switching power stage. Crossing-point estimates must be accurate in order to preserve audio quality. They must also be simple to calculate, in order to minimize processing requirements and delays. / We consider estimating the crossing points of a known function and a Gaussian random process, given uniformly-spaced, noisy samples of the random process for which the second-order statistics are assumed to be known. We derive the Maximum A-Posteriori (MAP) estimator, along with a Minimum Mean-Squared Error (MMSE) estimator which we show to be a computationally efficient approximation to the MAP estimator. / We also derive the Cramer-Rao bound (CRB) on estimator variance for the problem, which allows practical estimators to be evaluated against a best-case performance limit. We investigate several comparison estimators chosen from the literature. The structure of the MMSE estimator and comparison estimators is shown to be very similar, making the difference in computational expense between each technique largely dependent on the cost of evaluating various (generally non-linear) functions. / Simulations for both Pulse-Width and Click Modulation scenarios show the MMSE estimator performs very near to the Cramer-Rao bound and outperforms the alternative estimators selected from the literature.
220

High-speed analog-to-digital conversion in SiGe HBT technology

Li, Xiangtao 19 May 2008 (has links)
The objective of this research is to explore high-speed analog-to-digital converters (ADCs) using silicon-germanium (SiGe) heterojunction bipolar transistors (HBTs) for wireless digital receiver applications. The stringent requirements of ADCs for the high-performance next-generation wireless digital receiver include (1) low power, (2) low cost, (3) wide input signal bandwidth, (4) high sampling rate, and (5) medium to high resolution. The proposed research achieves the objective by implementing high-performance ADC's key building blocks and integrating these building blocks into a complete sigma-delta analog-to-digital modulator that satisfies the demanding specifications of next-generation wireless digital receiver applications. The scope of this research is divided into two main parts: (1) high-performance key building blocks of the ADC, and (2) high-speed sigma-delta analog-to-digital modulator. The research on ADC's building blocks includes the design of two high-speed track-and-hold amplifiers (THA) and two wide-bandwidth comparators operating at the sampling rate > 10 GS/sec with satisfying resolution. The research on high-speed sigma-delta analog-to-digital modulator includes the design and experimental characterization of a high-speed second-order low-pass sigma-delta modulator, which can operate with a sampling rate up to 20 GS/sec and with a medium resolution. The research is envisioned to demonstrate that the SiGe HBT technology is an ideal platform for the design of high-speed ADCs.

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