Showing papers on "Digital signal processing published in 1998"

Power consumption reduction in medical devices employing multiple digital signal processors

Abstract: Power consumption in medical devices is reduced through the use and operation of multiple digital signal processing systems. Each processor of the multiple systems performs at least one particular function in a predetermined time period. The multiple digital signal processors of such systems can be operated at lower clock frequencies relative to those that would be required by one of such processors to complete the multiple functions within the predetermined time period. With reduced clock frequency, power consumption is reduced. Further, with reduced clock speed, supply voltages applied to such digital signal processors may also be reduced.

Direct sequence spread spectrum method, computer-based product, apparatus and system tolerant to frequency reference offset

Abstract: A system employs a digital receiver (Fig. 9) (or transceiver) to receive, digitize and process a direct sequence spread spectrum signal using digital signal processing components. A radio front end portion (905) of the receiver receives and digitizes the signal (907), and a digital signal processing portion downconverts (909) and despreads (911) the signal by applying a pseudorandom noise (PN) code, used at a transmitter to spread a data signal contained in the direct sequence spread spectrum signal, to the received signal. A timing and state control mechanism (921) is included to provide time reference correction information to the signal processing components of the receiver for alignment of the PN code with the direct sequence spread spectrum signal, allowing the receiver to be compatible with transmitter using inaccurate frequency references which impart a significant frequency ambiguity in the received signal. Computer-based synchronization methods and mechanism suitable for use for low performance digital signal processors, and power management mechanisms (925) are employed to enable long-term operations using battery power, enabling utilization in network setting.

Computational signal processing with wavelets

Abstract: This unique resource examines the conceptual, computational, and practical aspects of applied signal processing using wavelets. With this book, readers will understand and be able to use the power and utility of new wavelet methods in science and engineering problems and analysis. The text is written in a clear, accessible style avoiding unnecessary abstractions and details. From a computational perspective, wavelet signal processing algorithms are presented and applied to signal compression, noise suppression, and signal identification. Numerical illustrations of these computational techniques are further provided with interactive software (MATLAB code) is available on the world wide web. Topics and Features: * Continuous wavelet and Gabor transforms * Frame-based theory of discretization and reconstruction of analog signals is developed * New and efficient "overcomplete" wavelet transform is introduced and applied * Numerical illustrations with an object-oriented computational perspective using the Wavelet Signal Processing Workstation (MATLAB code) available This book is an excellent resource for information and computational tools needed to use wavelets in many types of signal processing problems. Graduates, professionals, and practitioners in engineering, computer science, geophysics, and applied mathematics will benefit from using the book and software tools.

Wireless integrated network sensors: Low power systems on a chip

Abstract: Wireless Integrated Network Sensors (WINS) now provide a new monitoring and control capability for transportation, manufacturing, health care, environmental monitoring, and safety and security WINS combine sensing, signal processing, decision capability, and wireless networking capability in a compact, low power system WINS systems combine microsensor technology with low power sensor interface, signal processing, and RF communication circuits The need for low cost presents engineering challenges for implementation of these systems in conventional digital CMOS technology This paper describes micropower data converter, digital signal processing systems, and weak inversion CMOS RF circuits The digital signal processing system relies on a continuously operating spectrum analyzer Finally, the weak inversion CMOS RF systems are designed to exploit the properties of high-Q inductors to enable low power operation This paper reviews system architecture and low power circuits for WINS

Comparative study of peak power tracking techniques for solar storage system

Abstract: As the power supplied by solar arrays depends upon the insolation, temperature and array voltage, it is necessary to control the operating points to draw the maximum power of the solar array. The object of this paper is to investigate the maximum power tracking algorithms which were often used to compare the tracking efficiencies for the system operating under different controls. A simple method which combines a discrete time control and a PI compensator is used to track the maximum power points (MPPs) of the solar array. The implementation of the proposed converter system was based on a digital signal processor (DSP). The experimental tests were carried out, the tracking efficiencies are confirmed by simulations and experimental results.

Blind beamforming on a randomly distributed sensor array system

Abstract: We consider a digital signal processing sensor array system, based on randomly distributed sensor nodes, for surveillance and source localization applications. In most array processing the sensor array geometry is fixed and known and the steering array vector/manifold information is used in beamformation. In this system, array calibration may be impractical due to unknown placement and orientation of the sensors with unknown frequency/spatial responses. This paper proposes a blind beamforming technique, using only the measured sensor data, to form either a sample data or a sample correlation matrix. The maximum power collection criterion is used to obtain array weights from the dominant eigenvector associated with the largest eigenvalue of a matrix eigenvalue problem. Theoretical justification of this approach uses a generalization of Szego's (1958) theory of the asymptotic distribution of eigenvalues of the Toeplitz form. An efficient blind beamforming time delay estimate of the dominant source is proposed. Source localization based on a least squares (LS) method for time delay estimation is also given. Results based on analysis, simulation, and measured acoustical sensor data show the effectiveness of this beamforming technique for signal enhancement and space-time filtering.

Applications of digital signal processing to audio and acoustics

Abstract: With the advent of `multimedia', digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing to become a research field of its own. To date, most research in DSP applied to sound has been concentrated on speech, which is bandwidth limited to about 4 kilohertz. Speech processing is also limited by the low fidelity typically expected in the telephone network. Today, the main applications of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including perceptual measurement techniques and analysis/synthesis methods. Additional important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application-oriented research. The frequency range of wideband audio has an upper limit of 20 kilohertz and the resulting difference in frequency range and Signal to Noise Ratio (SNR) due to sample size must be taken into account when designing DSP algorithms. There are whole classes of algorithms that the speech community is not interested in pursuing or using. These algorithms and techniques are revealed in this book. This book is suitable for advanced level courses and serves as a valuable reference for researchers in the field. Interested and informed engineers will also find the book useful in their work.

Nonlinear Control of Electric Machinery

Abstract: From the Publisher: Establishing the mathematical foundation used throughout with an introduction on nonlinear control design and stability analysis, Nonlinear Control of Electric Machinery formulates appropriate control strategies for nonlinear dynamic models of electric machines ... details the most recent procedures for designing nonlinear algorithms ... validates the position tracking performance of 20 algorithms with test findings ... furnishes experimental and computer simulation results using actual motors and a PC-based digital signal processing system ... contains applications where various controllers are applied to direct-drive robot actuators ... and more.

Transducer assembly with smart connector

Abstract: A transducer assembly for connection with a digital signal processing system includes an analog transducer, a digital connector assembly movable relative to the analog transducer to facilitate connection with the digital signal processing system, and a cable permanently affixed between the analog transducer and the digital connector assembly to convey an analog transducer signal therebetween. The digital connector assembly includes a connector housing, a digital connector mounted by the connector housing to mate in a detachable manner with the digital signal processing system, and transducer interface circuitry disposed within the connector housing in a non-removable manner and including a digital storage device programmed to store digital transducer data, such as transducer identification, configuration settings and calibration or correction factors, for retrieval by the digital signal processing system. The transducer interface circuitry can also include signal conditioning circuitry and a microcontroller. Incorporating the transducer data memory and interface circuitry in the connector housing allows conventional transducers to be used without modifying existing mounting techniques and, at the same time, provides traceability of the transducer and its calibration data.

Adaptive frequency-domain equalization and diversity combining for broadband wireless communications

Abstract: We introduce a new kind of adaptive equalizer that operates in the spatial-frequency domain and uses either least mean square (LMS) or recursive least squares (RLS) adaptive processing. We simulate the equalizer's performance in an 8-Mb/s quaternary phase-shift keying (QPSK) link over a frequency-selective Rayleigh fading multipath channel with /spl sim/3 /spl mu/s RMS delay spread, corresponding to 60 symbols of dispersion. With the RLS algorithm and two diversity branches, our results show rapid convergence and channel tracking for a range of mobile speeds (up to /spl sim/100 mi/h). With a mobile speed of 40 mi/h, for example, the equalizer achieves an average bit error rate (BER) of 10/sup -4/ at a signal-to-noise ratio (SNR) of 15 dB, falling short of optimum linear receiver performance by about 4 dB. Moreover, it requires only /spl sim/50 complex operations per detected bit, i.e., /spl sim/400 M operations per second, which is close to achievable with state-of-the-art digital signal processing technology. An equivalent time-domain equalizer, if it converged at all, would require orders-of-magnitude more processing.

Fixed-point optimization utility for C and C++ based digital signal processing programs

Abstract: Fixed-point optimization utility software is developed that can aid scaling and wordlength determination of digital signal processing algorithms written in C or C++. This utility consists of two programs: the range estimator and the fixed-point simulator. The former estimates the ranges of floating-point variables for purposes of automatic scaling, and the latter translates floating-point programs into fixed-point equivalents to evaluate the fixed-point performance by simulation. By exploiting the operator overloading characteristics of C++, the range estimation and the fixed-point simulation can be conducted by simply modifying the variable declaration of the original program. This utility is easily applicable to nearly all types of digital signal processing programs including nonlinear, time-varying, multirate, and multidimensional signal processing algorithms. In addition, this software can be used to compare the fixed-point characteristics of different implementation architectures. An optimization example for an 8/spl times/8 inverse discrete cosine transform (IDCT) architecture that conforms to the IEEE standard specifications is presented. The optimized results require 8% fewer gates when compared with the previous best implementation.

Digital interference suppression system for radio frequency interference cancellation

Abstract: A digital signal processing system that produces an adaptive cancellation arrangement which nulls out all types of concurrent interference and/or jamming signals received by Global Positioning System (GPS) or spread spectrum receiver (7) from diverse antennas. In the present arrangement, orthogonal components of the composite received signal are separated by the receive antenna arrangement (3) and adjusted in the digital network (5) between the antenna (3) and the receiver (7) in phase and amplitude to optimally cancel components. The arrangements can be synergistically combined with digital adaptive transversal filter technology which is primarily used to supplement suppression performance by reducing narrowband interference in the band. The orthogonal received signal components from the GPS satellite constellation and from interference sources are combined in the present arrangement to adaptively create a null that attenuates interference sources while slightly modifying the GPS received signals.

Dynamic synapse for signal processing in neural networks

Abstract: An information processing system having signal processors that are interconnected by processing junctions that simulate and extend biological neural networks. As shown in the figure, each processing junction receives signals from one signal processor and generates a new signal to another signal processor. The response of each processing junction is determined by internal junction processes and is continuously changed with temporal variation in the received signal. Different processing junctions connected to receive a common signal from a signal processor respond differently to produce different signals to downstream signal processors. This transforms a temporal pattern of a signal train of spikes into a spatio-temporal pattern of junction events and provides an exponential computational power to signal processors. Each signal processing junction can receive a feedback signal from a downstream signal processor so that an internal junction process can be adjusted to learn certain characteristics embedded in received signals.

A continuously calibrated 12-b, 10-MS/s, 3.3-V A/D converter

Abstract: The continuous calibration of high-linearity, highspeed analog/digital converters (ADCs) can minimize system complexity by allowing a single converter to maintain its accuracy over time. This paper introduces a continuous calibration technique for pipelined and successive approximation ADCs that avoids some of the limitations of earlier designs by performing the calibration in the analog domain. The calibration is made transparent to the overall system by employing an extra stage that is calibrated outside of the main converter's operation and periodically substituted for a stage within the main converter. A 12-b, pipelined ADC employing this architecture has been integrated in a 0.5-/spl mu/m, single-poly, quadruple-metal, 3.3-V CMOS technology. The measured dynamic performance indicates that at a 10-MHz sampling rate, the circuit achieves a peak signal-to-noise-plus-distortion ratio of 67 dB and a total harmonic distortion of -77 dR for a 4.8-MHz input. The total power dissipated by the prototype is 335 mW, and its active area is 3.71/spl times/3.91 mm/sup 2/.

Architectures for Digital Signal Processing

Abstract: From the Publisher: Digital Signal Processing is a rapidly expanding area for evaluation and development of efficient measures for representation, transformation and manipulation of signals. This book fuses signal processing algorithms and VLSI circuit design to assist digital signal processing architecture developers. The author also shows how this technique can be used in applications such as: signal transmission and storage, manufacturing process quality control and assurance, autonomous mobile system control and biomedical process analysis.

Active noise cancellation aircraft headset system

Abstract: An active noise cancellation aircraft headset system. A speaker is mounted within each earcup of a headset for receiving and acoustically transducing a composite noise cancellation signal. A microphone is also mounted within each earcup for transducing acoustic pressure within the earcup to a corresponding analog error signal. An analog filter receives the analog error signal and inverts it to generate an analog broadband noise cancellation signal. The analog error signal is also provided to an analog to digital converter, which receives the analog microphone error signal and converts it to a digital error signal. A DSP takes the digital error signal and, using an adaptive digital feedback filter, generates a digital tonal noise cancellation signal. A digital to analog converter then converts the digital tonal noise cancellation signal to an analog tonal noise cancellation signal so that it can be combined with the analog broadband noise cancellation signal. The resultant composite cancellation signal is provided to the speakers in the earcups to cancel noise within the earcups. The broadband analog cancellation is effective to reduce overall noise within the earcup, and the DSP not only provides active control of the analog cancellation loop gain to maximize the effectiveness of the broadband analog cancellation but also uses the adaptive feedback filter/algorithm to substantially reduce at least the loudest tonal noises penetrating the earcup, including engine and propeller noises, as well as harmonic vibrations of components of the aircraft's fuselage.

A binaural digital hearing aid system

Abstract: A binaural digital hearing aid system comprises two hearing aid units (1, 2) for arrangement in a user's left and right ear, respectively. Each unit comprises input signal transducer means (3r, 3l), A/D conversion means (4r, 4l), digital signal processing means (5r-13r, 5l-13l), D/A conversion means (14r, 14l) and output signal transducer means (15r, 15l). A bi-directional communication link (7) is provided between the units. The digital signal processing means of each unit is arranged to affect a substantially full digital signal processing including individual processing of signals from the input transducer means of the actual unit and simulated processing of signals from the input transducer means of the other unit as well as binaural signal processing and includes at least a first digital signal processor part (5r, 5l) for processing said internally supplied signal, a second digital signal processor part (6l, 6r) for processing the signal supplied via said communication link (7) and a third digital signal processor part (9r, 9l) to effect common binaural digital signal processing of information derived from the signals processed in said first and second digital signal processor parts, said second digital signal processor part (6l, 6r) in each unit simulating the first digital signal processor part (5l, 5r) in the other unit with respect to adjustment parameters controlling the performance of said first signal processor part in said other unit.

Apparatus for and method of filtering in an digital hearing aid, including an application specific integrated circuit and a programmable digital signal processor

Abstract: An apparatus for a hearing aid provides an application specific integrated circuit (ASIC) for filtering of input signals and a programmable digital signal processor connected to it, for control of filterbank gains. This provides a compromise between the conflicting goals of size, re-programmability and power consumption. The fixed portion of the processing, i.e. filtering is implemented in hardware in the ASIC, and the variable or adjustable portion of the processing is implemented by the programmable digital signal processor. The filterbank has an adjustable number of channels and means for shifting the center frequencies of the bands in unison to one of two discrete sets of center frequencies. A wide range of hearing loss compensation schemes can be implemented. Software programs can be executed on the programmable digital signal processor.

Nonlinear and nonstationary signal processing

Abstract: Presents a brief discussion of the need for and evolution to nonlinear and nonstationary signal processing. Applications where these are useful are mentioned.

A field programmable radio frequency communications equipment including a configurable if circuit and method therefor

Abstract: A system and method for providing a field configurable radio frequency communications system including an intermediate frequency digital signal processing circuit therefor. The intermediate frequency digital signal processing circuits includes digital transmitter and receiver circuit paths, a digital baseband circuit, control registers and an interface circuit. The transmitter and receiver circuit paths and the baseband circuits include digital processing circuits that are configurable by commands provided to the control register via the interface circuit to select the transmitter or receiver mode of operation, to select the signaling scheme of waveform employed, to configure the signal processing functions of the transmitter or receiver circuit paths, and to select the use of any of a plurality of a baseband signal processing functions.

An Extensible Real-time Signal Processing Environment for Max

Abstract: MSP is a set of additions to the Macintosh version of MAX for synthesis, signal processing, sampling, and harddisk recording/playback directly on a PowerPC processor. It is based on ideas from MAX/FTS developed by Miller Puckette at IRCAM, and incorporates software from Puckette's cross-platform Pd (Pure Data) program.

Flexible, modular electronic element patterning method and apparatus for compiling, processing, transmitting, and reporting data and information

Abstract: A method and apparatus for compiling (acquiring and storing), processing (analyzing, integrating, and organizing), transmitting, and reporting data and information, which is comprised of a flexible, modular system that overcomes major limitations of conventional and multidimensional databases. A method and apparatus that utilizes computer programming code modules (601) and modules of logically arranged digital signal function and formula formations to: (a) acquire data/information units (602) using modules of query instruction items and response instruction items (603), which can be presented via a branching-logic process, (b) store the responses to the items in independent record files and internal database files, (c) integrate them with digital signals stored in other source (605) via an integration file, (d) process the digital signals in the digital signal processing files, (e) produce portable report files (608), and (f) generate reports utilizing report format files (609).

Design of a fibre optic multi-point sensor for gas detection

Abstract: We report the design of a multi-point fibre optic methane sensor using a DFB laser source with a branched fibre network and micro-optic cells. Measurements are performed through derivative spectroscopy, with line scanning and digital signal processing, to give sensitivities down to a few ppm metre. The form of the derivative signal obtained from the system is modelled theoretically and compared with the experimental signal. The main limitation in the signal to noise ratio of the system is due to interference effects (etalon fringes) from the cells and we show how these effects may be minimised.

DSP processors hit the mainstream

Abstract: These days, the once obscure engineering term "DSP" (digital signal processing) is working its way into common use. It has begun to crop up on the labels of an ever wider range of products, from home audio components to answering machines. This is not merely a reflection of a new marketing strategy, however; there truly is more digital signal processing inside today's products than ever before. But why is the market for DSP processors booming? The answer is somewhat circular: as microprocessor fabrication processes have become more sophisticated, the cost of a microprocessor capable of performing DSP tasks has dropped significantly to the point where such a processor can be used in consumer products and other cost sensitive systems. As a result, more and more products have begun using DSP processors, fueling demand for faster, smaller, cheaper, more energy-efficient chips. Although fundamentally related, DSP processors are significantly different from general purpose processors (GPPs) like the Intel Pentium or PowerPC. The authors explain what DSP processors are and what they do. They also offer a guide to evaluating DSP processors for use in a product or application.

Performance analysis of technology using the S curve model: the case of digital signal processing (DSP)

Abstract: The purpose of this paper is to analyse the evolution of the technology performance of Digital Signal Processing components (DSPs) using the S curve model. In the first part, the theoretical base of this model is established through a comparative study between the S curve model and other concepts with which it is closely related: innovation diffusion models and life cycle models. The purpose of this study is to use the solid analytical foundation of these models to increase the theoretical consistency of the S curve model. In the second part of the article, a methodology that facilitates the application of this model is proposed. At the same time, the usefulness of the S curve as a strategic analysis tool is discussed as well as the problems that can arise when the model is put into practice. This methodology is used in the analysis of the technology of DSPs. © 1998 Elsevier Science Ltd. All rights reserved

Advanced Video-Based Surveillance Systems

Abstract: From the Publisher: Advanced Video-Based Surveillance Systems presents second generation surveillance systems that automatically process large sets of signals for performance monitoring tasks. Included is coverage of different architecture designs, customization of surveillance architecture for end-users, advances in the processing of imaging sequences, security systems, sensors, and remote monitoring projects. Examples are provided of surveillance applications in highway traffic control, subway stations, wireless communications, and other areas. This work will be of interest to researchers in image processing, computer vision, digital signal processing, and telecommunications.

Evaluating MMX technology using DSP and multimedia applications

Abstract: Many current general purpose processors are using extensions to the instruction set architecture to enhance the performance of digital signal processing (DSP) and multimedia applications. In this paper, we evaluate the X86 architecture's multimedia extension (MMX) instruction set on a set of benchmarks. Our benchmark suite includes kernels (filtering, fast Fourier transforms, and vector arithmetic) and applications (JPEG compression, Doppler radar processing, imaging, and G.722 speech encoding). Each benchmark has at least one non-MMX version in C and an MMX version that makes calls to an MMX assembly library. The versions differ in the implementation of filtering, vector arithmetic, and other relevant kernels. The observed speed up for the MMX versions of the suite ranges from less than 1.0 to 6.1. In addition to quantifying the speedup, we perform detailed instruction level profiling using Intel's VTune profiling tool. Using VTune, we profile static and dynamic instructions, microarchitecture operations, and data references to isolate the specific reasons for speedup or lack thereof. This analysis allows one to understand which aspects of native signal processing instruction sets are most useful, the current limitations, and how they can be utilized most efficiently.

Power amplifier arrangement and method for data signal interface

Abstract: An arrangement and method using power more efficiently for signal amplification. The approach is particularly advantageous in application environments where digital information is converted to the analog domain and carried at a variety of different magnitudes. One embodiment involves a signal processing and amplifying arrangement including a digital-to-analog converter, an amplifier and a control signal selection circuit. The digital-to-analog converter receives a digital signal and outputs, in response to the digital signal, an analog signal. The amplifier amplifies the analog signal using power received from a high-level supply and a low-level supply, at least one of which has a variable setting in response to a control signal. The amplifier is configured and arranged to amplify the analog signal. The control signal selection circuit is responsive to the digital signal, and is used for selecting the control signal to cause the supplies to the amplifier to track the expected magnitude of the analog signal.

Optical interface between receiver and tag response signal analyzer in RFID system for detecting low power resonant tags

Abstract: Noise reduction schemes are provided in a radio frequency identification (RFID) system for use with RFID intelligent tags. Fiber optics are used to communicate analog tag response signals from the output of the receiver circuit to the input of a tag response signal analyzing module, which includes a digital signal processor (DSP). The fiber optics creates electrical isolation between these circuit elements breaking ground loops, stopping internal switching noise from the DSP from entering the receiver circuitry, and preventing common mode signals from interfering with the desired RFID tag signal.