Efficient Low Power Headphone Driver

In recent years, the consumer electronics market for battery-powered devices such as smartphones and tablets has been rapidly expanding. The requirements for audio CODEC in these portable devices have extended from merely supporting voice calls to high-fidelity music playback. As a result, audio driver performance has become one of the most important differentiating factors among products from different suppliers. There are three basic performance metrics that are typically used to benchmark audio modules: the maximum delivered output power, the audio fidelity measured in terms of dynamic range, THD+N, and finally the battery life. Maximizing all three of these performance metrics has proven to be an exceptionally hard task as portrayed by the research publications.This work presents an attempt to push all three of these metrics together and provide an acceptable balance which is achieved by selecting the right topology. Conventionally, headphone drivers are designed using a linear amplifier topology for many reasons- most prominently- to achieve a superior THD+N and PSRR requirement which in the past was essentially the only key performance metric needed. This came at the expense of realizing mediocre power efficiency targets, thereby wasting battery life. This picture changed dramatically over the last decade with smartphones and other portable devices becoming the first choice of the young generation. These devices are extremely power hungry due to the unlimited functions and features they provide and therefore battery life has come to the spotlight as a key resource that need to be preserved. As a result, in this work a headphone driver is based on a switching topology that is able to deliver more than 230mW of power (or equivalently 2Vrms) to a 16Ω load while achieving better than -98dB of THD+N , more than 108dB of SNR, and about 108dB PSRR while still maintaining a peak power efficiency of more than 84%

Recent advances in the hardware architecture of flat display devices

Thesis (Master)--Izmir Institute of Technology, Electronics and Communication Engineering, Izmir, 2007Includes bibliographical References (leaves: 115-117)Text in English; Abstract: Turkish and Englishxiii, 133 leavesThesis will describe processing board hardware design for flat panel displays with integrated digital reception, the design challenges in flat panel displays with integrated digital reception explained with details. Thesis also includes brief explanation of flat panel technology and processing blocks. Explanations of building blocks of TV and flat panel displays are given before design stage for better understanding of design stage. Hardware design stage of processing board is investigated in two major steps, schematic design and layout design. First step of the schematic design is system level block diagram design. Schematic diagram is the detailed application level hardware design and layout is the implementation level of the design. System level, application level and implementation level hardware design of the TV processing board is described with details in thesis. Design challenges, considerations and solutions are defined in advance for flat panel displays

Low Power High Efficiency Integrated Class-D Amplifier Circuits for Mobile Devices

The consumer’s demand for state-of-the-art multimedia devices such as smart phones and tablet computers has forced manufacturers to provide more system features to compete for a larger portion of the market share. The added features increase the power consumption and heat dissipation of integrated circuits, depleting the battery charge faster. Therefore, low-power high-efficiency circuits, such as the class-D audio amplifier, are needed to reduce heat dissipation and extend battery life in mobile devices. This dissertation focuses on new design techniques to create high performance class-D audio amplifiers that have low power consumption and occupy less space. The first part of this dissertation introduces the research motivation and fundamentals of audio amplification. The loudspeaker’s operation and main audio performance metrics are examined to explain the limitations in the amplification process. Moreover, the operating principle and design procedure of the main class-D amplifier architectures are reviewed to provide the performance tradeoffs involved. The second part of this dissertation presents two new circuit designs to improve the audio performance, power consumption, and efficiency of standard class-D audio amplifiers. The first work proposes a feed-forward power-supply noise cancellation technique for single-ended class-D amplifier architectures to improve the power-supply rejection ratio across the entire audio frequency range. The design methodology, implementation, and tradeoffs of the proposed technique are clearly delineated to demonstrate its simplicity and effectiveness. The second work introduces a new class-D output stage design for piezoelectric speakers. The proposed design uses stacked-cascode thick-oxide CMOS transistors at the output stage that makes possible to handle high voltages in a low voltage standard CMOS technology. The design tradeoffs in efficiency, linearity, and electromagnetic interference are discussed. Finally, the open problems in audio amplification for mobile devices are discussed to delineate the possible future work to improve the performance of class-D amplifiers. For all the presented works, proof-of-concept prototypes are fabricated, and the measured results are used to verify the correct operation of the proposed solutions

Study and design of an interface for remote audio processing

This project focused on the study and design of an interface for remote audio processing, with the objective of acquiring by filtering, biasing, and amplifying an analog signal before digitizing it by means of two MCP3208 ADCs to achieve a 24-bit resolution signal. The resulting digital signal was then transmitted to a Raspberry Pi using SPI protocol, where it was processed by a Flask server that could be accessed from both local and remote networks. The design of the PCB was a critical component of the project, as it had to accommodate various components and ensure accurate signal acquisition and transmission. The PCB design was created using KiCad software, which allowed for the precise placement and routing of all components. A major challenge in the design of the interface was to ensure that the analog signal was not distorted during acquisition and amplification. This was achieved through careful selection of amplifier components and using high-pass and low-pass filters to remove any unwanted noise. Once the analog signal was acquired and digitized, the resulting digital signal was transmitted to the Raspberry Pi using SPI protocol. The Raspberry Pi acted as the host for a Flask server, which could be accessed from local and remote networks using a web browser. The Flask server allowed for the processing of the digital signal and provided a user interface for controlling the gain and filtering parameters of the analog signal. This enabled the user to adjust the signal parameters to suit their specific requirements, making the interface highly flexible and adaptable to a variety of audio processing applications. The final interface was capable of remote audio processing, making it highly useful in scenarios where the audio signal needed to be acquired and processed in a location separate from the user. For example, it could be used in a recording studio, where the audio signal from the microphone could be remotely processed using the interface. The gain and filtering parameters could be adjusted in real-time, allowing the sound engineer to fine-tune the audio signal to produce the desired recording. In conclusion, the project demonstrated the feasibility and potential benefits of using a remote audio processing system for various applications. The design of the PCB, selection of components, and use of the Flask server enabled the creation of an interface that was highly flexible, accurate, and adaptable to a variety of audio processing requirements. Overall, the project represents a significant step forward in the field of remote audio processing, with the potential to benefit many different applications in the future

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

A new stethoscope for reduction of heart sounds from lung sound recordings.

Yip Lung.Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.Includes bibliographical references.Abstracts in English and Chinese.Chapter 1 --- IntroductionChapter 1.1 --- Heart and Lung Diseases --- p.1Chapter 1.1.1 --- Hong Kong --- p.1Chapter 1.1.2 --- China --- p.2Chapter 1.1.3 --- the United States of America (USA) --- p.3Chapter 1.2 --- Auscultation --- p.3Chapter 1.2.1 --- Introduction of Auscultation --- p.4Chapter 1.2.2 --- Comparison between Auscultation and Ultrasound --- p.6Chapter 1.3 --- Stethoscope --- p.7Chapter 1.3.1 --- History of Stethoscope --- p.7Chapter 1.3.2 --- New Electronic Stethoscope --- p.14Chapter 1.4 --- Main Purpose of the Study --- p.16Chapter 1.5 --- Organization of Thesis --- p.16References --- p.18Chapter 2 --- A New Electronic Stethoscope's HeadChapter 2.1 --- Introduction --- p.20Chapter 2.2 --- Biopotential Electrode --- p.21Chapter 2.2.1 --- Flexible Electrode --- p.21Chapter 2.2.2 --- Laplacian Electrocardiogram --- p.22Chapter 2.3 --- Transducer --- p.25Chapter 2.4 --- Design of the Head of Stethoscope --- p.26Chapter 2.5 --- Experimental Results --- p.27Chapter 2.5.1 --- Bias Voltage of Condenser Microphone --- p.27Chapter 2.5.2 --- Frequency Response of New Stethoscope's Head --- p.29Chapter 2.6 --- Discussion --- p.30Chapter 2.7 --- Section Summary --- p.31References --- p.33Chapter 3 --- Signal Pre-processing UnitChapter 3.1 --- Introduction --- p.35Chapter 3.2 --- High Input Impedance IC Amplifier --- p.36Chapter 3.3 --- Voltage Control Voltage Source High Pass Filter Circuit --- p.37Chapter 3.4 --- Multiple Feed Back Low Pass Filter Circuit --- p.39Chapter 3.5 --- Overall Circuit --- p.41Chapter 3.6 --- Experimental Results --- p.43Chapter 3.7 --- Discussion --- p.46Chapter 3.8 --- Section Summary --- p.47References --- p.48Chapter 4 --- Central PlatformChapter 4.1 --- Introduction --- p.49Chapter 4.2 --- Adaptive Filter --- p.49Chapter 4.2.1 --- Introduction to Adaptive Filtering --- p.49Chapter 4.2.2 --- Least-Mean-Square (LMS) Algorithm --- p.51Chapter 4.2.3 --- Applications --- p.52Chapter 4.3 --- Offline Processing --- p.54Chapter 4.3.1 --- WINDAQ and MATLAB --- p.55Chapter 4.3.2 --- Direct Reference Algorithm --- p.57Chapter 4.3.3 --- Determination of Parameters in DRA --- p.62Chapter 4.3.4 --- Experimental Results of DRA --- p.67Chapter 4.3.5 --- Acoustic Waveform Based Algorithm --- p.72Chapter 4.3.6 --- Experimental Results of AWBA --- p.81Chapter 4.4 --- Online Processing --- p.85Chapter 4.4.1 --- LABVIEW --- p.85Chapter 4.4.2 --- Automated Gain Control --- p.88Chapter 4.4.3 --- Implementation of LMS adaptive filter --- p.89Chapter 4.4.4 --- Experimental Results of Online-AGC --- p.92Chapter 4.5 --- Discussion --- p.93Chapter 4.6 --- Section Summary --- p.97References --- p.98Chapter 5 --- Conclusion and Further DevelopmentChapter 5.1 --- Conclusion of the Main Contribution --- p.100Chapter 5.2 --- Future Works --- p.102Chapter 5.2.1 --- Modification of the Head of Stethoscope --- p.102Chapter 5.2.2 --- Validation of Abnormal Breath --- p.102Chapter 5.2.3 --- Low Frequency Analysis --- p.102Chapter 5.2.4 --- AGC-AWBA Approach --- p.102Chapter 5.2.5 --- Standalone Device --- p.103Chapter 5.2.6 --- Demand on Stethoscope --- p.109References --- p.110AppendixChapter A.1 --- Determination of parameters in VCVS High Pass Filter --- p.106Chapter A.2 --- Determination of parameters in MFB Low Pass Filter --- p.110Chapter A.3 --- Source code of DRA (MATLAB) --- p.114Chapter A.4 --- Source code of AWBA (MATLAB) --- p.129Chapter A.5 --- Source code of online AGC (LABVIEW) --- p.13

Ultra-high-speed imaging of bubbles interacting with cells and tissue

Ultrasound contrast microbubbles are exploited in molecular imaging, where bubbles are directed to target cells and where their high-scattering cross section to ultrasound allows for the detection of pathologies at a molecular level. In therapeutic applications vibrating bubbles close to cells may alter the permeability of cell membranes, and these systems are therefore highly interesting for drug and gene delivery applications using ultrasound. In a more extreme regime bubbles are driven through shock waves to sonoporate or kill cells through intense stresses or jets following inertial bubble collapse. Here, we elucidate some of the underlying mechanisms using the 25-Mfps camera Brandaris128, resolving the bubble dynamics and its interactions with cells. We quantify acoustic microstreaming around oscillating bubbles close to rigid walls and evaluate the shear stresses on nonadherent cells. In a study on the fluid dynamical interaction of cavitation bubbles with adherent cells, we find that the nonspherical collapse of bubbles is responsible for cell detachment. We also visualized the dynamics of vibrating microbubbles in contact with endothelial cells followed by fluorescent imaging of the transport of propidium iodide, used as a membrane integrity probe, into these cells showing a direct correlation between cell deformation and cell membrane permeability

Aerospace medicine and biology: A cumulative index to a continuing bibliography (supplement 384)

This publication is a cumulative index to the abstracts contained in Supplements 372 through 383 of Aerospace Medicine and Biology: A Continuing Bibliography. It includes seven indexes: subject, personal author, corporate source, foreign technology, contract number, report number, and accession number