On chopper effects in discrete-time ΣΔ modulators

Analog-to-digital converters based on ΣΔ modulators are used in a wide variety of applications. Due to their inherent monotonous behavior, high linearity, and large dynamic range, they are often the preferred option for sensor and instrumentation. Offset and flicker noise are usual concerns for this type of applications, and one way to minimize their effects is to use a chopper in the front-end integrator of the modulator. Due to its simple operation principle, the action of the chopper in the integrator is often overlooked. In this paper, we provide an analytical study of the static effects in ΣΔ modulators, which shows that the introduction of chopper is not transparent to the modulator operation and should thus be designed with care.Gobierno de España TEC-2007-68072Consejo Superior de Investigaciones Científicas 200850I21

On chopper effects in discrete-time ΣΔ modulators

Analog-to-digital converters based on ΣΔ modulators are used in a wide variety of applications. Due to their inherent monotonous behavior, high linearity, and large dynamic range, they are often the preferred option for sensor and instrumentation. Offset and flicker noise are usual concerns for this type of applications, and one way to minimize their effects is to use a chopper in the front-end integrator of the modulator. Due to its simple operation principle, the action of the chopper in the integrator is often overlooked. In this paper, we provide an analytical study of the static effects in ΣΔ modulators, which shows that the introduction of chopper is not transparent to the modulator operation and should thus be designed with care.This work has been partially funded by the Spanish Government project TEC-2007-68072 and the CSIC project 200850I213.Peer reviewe

Improved design techniques for low-voltage low-power switched-capacitor delta-sigma modulators

This dissertation investigates the constraints which arise when switched-capacitor (SC) delta-sigma modulators are designed for low-voltage operation, targeting also low power dissipation, and proposes methods of improving the performance and optimizing for low power dissipation. This is accomplished by identifying critical elements whose performance can lead to increased power dissipation, as well as the fundamental limitations of available analog circuit techniques. A prototype was designed and fabricated, which reflected these findings, and therefore exhibited good performance and nearly optimum power dissipation. One of the key performance parameters is the dc gain of the amplifier in the first stage; it should be high. This is necessary for high linearity and low quantization noise leakage. In low-voltage operation, it may become impractical to use conventional topologies employing cascoding techniques (e.g., folded-cascode) which provide high gain in one single stage. Rather, cascaded structures have to be used. The disadvantage of the latter is the necessity for frequency compensation which results in increased power dissipation. Hence, another objective of this work is to exploit techniques which compensate for the open-loop gain characteristic of the amplifier (dc gain and nonlinearity), thus permitting the utilization of single-stage low-gain topologies. Predictive correlated double sampling is one of such techniques and is analyzed in detail

Efficient compression of motion compensated residuals

EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Analysis and resynthesis of polyphonic music

This thesis examines applications of Digital Signal Processing to the analysis, transformation, and resynthesis of musical audio. First I give an overview of the human perception of music. I then examine in detail the requirements for a system that can analyse, transcribe, process, and resynthesise monaural polyphonic music. I then describe and compare the possible hardware and software platforms. After this I describe a prototype hybrid system that attempts to carry out these tasks using a method based on additive synthesis. Next I present results from its application to a variety of musical examples, and critically assess its performance and limitations. I then address these issues in the design of a second system based on Gabor wavelets. I conclude by summarising the research and outlining suggestions for future developments

Algorithms and VLSI architectures for parametric additive synthesis

A parametric additive synthesis approach to sound synthesis is advantageous as it can model sounds in a large scale manner, unlike the classical sinusoidal additive based synthesis paradigms. It is known that a large body of naturally occurring sounds are resonant in character and thus fit the concept well. This thesis is concerned with the computational optimisation of a super class of form ant synthesis which extends the sinusoidal parameters with a spread parameter known as band width. Here a modified formant algorithm is introduced which can be traced back to work done at IRCAM, Paris. When impulse driven, a filter based approach to modelling a formant limits the computational work-load. It is assumed that the filter's coefficients are fixed at initialisation, thus avoiding interpolation which can cause the filter to become chaotic. A filter which is more complex than a second order section is required. Temporal resolution of an impulse generator is achieved by using a two stage polyphase decimator which drives many filterbanks. Each filterbank describes one formant and is composed of sub-elements which allow variation of the formant’s parameters. A resource manager is discussed to overcome the possibility of all sub- banks operating in unison. All filterbanks for one voice are connected in series to the impulse generator and their outputs are summed and scaled accordingly. An explorative study of number systems for DSP algorithms and their architectures is investigated. I invented a new theoretical mechanism for multi-level logic based DSP. Its aims are to reduce the number of transistors and to increase their functionality. A review of synthesis algorithms and VLSI architectures are discussed in a case study between a filter based bit-serial and a CORDIC based sinusoidal generator. They are both of similar size, but the latter is always guaranteed to be stable

Design and Control of Power Converters 2019

In this book, 20 papers focused on different fields of power electronics are gathered. Approximately half of the papers are focused on different control issues and techniques, ranging from the computer-aided design of digital compensators to more specific approaches such as fuzzy or sliding control techniques. The rest of the papers are focused on the design of novel topologies. The fields in which these controls and topologies are applied are varied: MMCs, photovoltaic systems, supercapacitors and traction systems, LEDs, wireless power transfer, etc

Low Power Digital Filter Implementation in FPGA

Digital filters suitable for hearing aid application on low power perspective have been developed and implemented in FPGA in this dissertation. Hearing aids are primarily meant for improving hearing and speech comprehensions. Digital hearing aids score over their analog counterparts. This happens as digital hearing aids provide flexible gain besides facilitating feedback reduction and noise elimination. Recent advances in DSP and Microelectronics have led to the development of superior digital hearing aids. Many researchers have investigated several algorithms suitable for hearing aid application that demands low noise, feedback cancellation, echo cancellation, etc., however the toughest challenge is the implementation. Furthermore, the additional constraints are power and area. The device must consume as minimum power as possible to support extended battery life and should be as small as possible for increased portability. In this thesis we have made an attempt to investigate possible digital filter algorithms those are hardware configurable on low power view point. Suitability of decimation filter for hearing aid application is investigated. In this dissertation decimation filter is implemented using ‘Distributed Arithmetic’ approach.While designing this filter, it is observed that, comb-half band FIR-FIR filter design uses less hardware compared to the comb-FIR-FIR filter design. The power consumption is also less in case of comb-half band FIR-FIR filter design compared to the comb-FIR-FIR filter. This filter is implemented in Virtex-II pro board from Xilinx and the resource estimator from the system generator is used to estimate the resources. However ‘Distributed Arithmetic’ is highly serial in nature and its latency is high; power consumption found is not very low in this type of filter implementation. So we have proceeded for ‘Adaptive Hearing Aid’ using Booth-Wallace tree multiplier. This algorithm is also implemented in FPGA and power calculation of the whole system is done using Xilinx Xpower analyser. It is observed that power consumed by the hearing aid with Booth-Wallace tree multiplier is less than the hearing aid using Booth multiplier (about 25%). So we can conclude that the hearing aid using Booth-Wallace tree multiplier consumes less power comparatively. The above two approached are purely algorithmic approach. Next we proceed to combine circuit level VLSI design and with algorithmic approach for further possible reduction in power. A MAC based FDF-FIR filter (algorithm) that uses dual edge triggered latch (DET) (circuit) is used for hearing aid device. It is observed that DET based MAC FIR filter consumes less power than the traditional (single edge triggered, SET) one (about 41%). The proposed low power latch provides a power saving upto 65% in the FIR filter. This technique consumes less power compared to previous approaches that uses low power technique only at algorithmic abstraction level. The DET based MAC FIR filter is tested for real-time validation and it is observed that it works perfectly for various signals (speech, music, voice with music). The gain of the filter is tested and is found to be 27 dB (maximum) that matches with most of the hearing aid (manufacturer’s) specifications. Hence it can be concluded that FDF FIR digital filter in conjunction with low power latch is a strong candidate for hearing aid application

Digital Filters

The new technology advances provide that a great number of system signals can be easily measured with a low cost. The main problem is that usually only a fraction of the signal is useful for different purposes, for example maintenance, DVD-recorders, computers, electric/electronic circuits, econometric, optimization, etc. Digital filters are the most versatile, practical and effective methods for extracting the information necessary from the signal. They can be dynamic, so they can be automatically or manually adjusted to the external and internal conditions. Presented in this book are the most advanced digital filters including different case studies and the most relevant literature