Showing papers on "Digital signal processing published in 1999"

Splines: a perfect fit for signal and image processing

Abstract: The article provides arguments in favor of an alternative approach that uses splines, which is equally justifiable on a theoretical basis, and which offers many practical advantages. To reassure the reader who may be afraid to enter new territory, it is emphasized that one is not losing anything because the traditional theory is retained as a particular case (i.e., a spline of infinite degree). The basic computational tools are also familiar to a signal processing audience (filters and recursive algorithms), even though their use in the present context is less conventional. The article also brings out the connection with the multiresolution theory of the wavelet transform. This article attempts to fulfil three goals. The first is to provide a tutorial on splines that is geared to a signal processing audience. The second is to gather all their important properties and provide an overview of the mathematical and computational tools available; i.e., a road map for the practitioner with references to the appropriate literature. The third goal is to give a review of the primary applications of splines in signal and image processing.

Active noise control: a tutorial review

Abstract: Active noise control (ANC) is achieved by introducing a cancelling "antinoise" wave through an appropriate array of secondary sources. These secondary sources are interconnected through an electronic system using a specific signal processing algorithm for the particular cancellation scheme. ANC has application to a wide variety of problems in manufacturing, industrial operations, and consumer products. The emphasis of this paper is on the practical aspects of ANC systems in terms of adaptive signal processing and digital signal processing (DSP) implementation for real-world applications. In this paper, the basic adaptive algorithm for ANC is developed and analyzed based on single-channel broad-band feedforward control. This algorithm is then modified for narrow-band feedforward and adaptive feedback control. In turn, these single-channel ANC algorithms are expanded to multiple-channel cases. Various online secondary-path modeling techniques and special adaptive algorithms, such as lattice, frequency-domain, subband, and recursive-least-squares, are also introduced. Applications of these techniques to actual problems are highlighted by several examples.

DSP integrated circuits

Abstract: DSP Integrated Circuits. VLSI Circuit Technologies. Digital Signal Processing. Digital Filters. Finite Word Length Effects. DSP Algorithms. DSP System Design. Architectures for DSP. Synthesis of DSP Architectures. Digital Systems. Processing Elements. Integrated Circuit Design. Subject Index.

Synthesis of Embedded Software from Synchronous Dataflow Specifications

Abstract: The implementation of software for embedded digital signal processing (DSP) applications is an extremely complex process. The complexity arises from escalating functionality in the applicationss intense time-to-market pressuress and stringent cost, power and speed constraints. To help cope with such complexity, DSP system designers have increasingly been employing high-level, graphical design environments in which system specification is based on hierarchical dataflow graphs. Consequently, a significant industry has emerged for the development of data-flow-based DSP design environments. Leading products in this industry include SPW from Cadence, COSSAP from Synopsys, ADS from Hewlett Packard, and DSP Station from Mentor Graphics. This paper reviews a set of algorithms for compiling dataflow programs for embedded DSP applications into efficient implementations on programmable digital signal processors. The algorithms focus primarily on the minimization of code size, and the minimization of the memory required for the buffers that implement the communication channels in the input dataflow graph. These are critical problems because programmable digital signal processors have very limited amounts of on-chip memory, and the speed, power, and cost penalties for using off-chip memory are often prohibitively high for embedded applications. Furthermore, memory demands of applications are increasing at a significantly higher rate than the rate of increase in on-chip memory capacity offered by improved integrated circuit technology.

The digital front-end of software radio terminals

Abstract: When expanding digital signal processing of mobile communications terminals toward the antenna while making the terminal more wideband in order to be able to cope with different mobile communications standards in a software radio based terminal, the designer is faced with strong requirements such as bandwidth and dynamic range. Many publications claim that only reconfigurable hardware such as FPGAs can simultaneously cope with such diversity and requirements. Starting with considerations of the receiver architecture, we describe key functionalities of the digital front-end and highlight how the signal characteristics of mobile communications signals and commonalities among different signal processing operations can be exploited to great advantage, eventually enabling implementations on an ASIC that, although not reconfigurable, would empower the software radio concept.

Hand held ultrasonic diagnostic instrument

Abstract: A hand held ultrasonic instrument (87) is provided in a portable unit which performs both B mode and Doppler imaging. The instrument includes a transducer array (10) mounted in a hand-held enclosure, with an integrated circuit transceiver connected to the elements of the array (10) for the reception of echo signals. A digital signal processing circuit (40) performs both B mode, and Doppler estimation, as well as advanced functions such as assembly of multiple zone focused scan lines, synthetic aperture formation, depth dependent filtering, speckle reduction, flash suppression, and frame averaging.

Real-time on-line test for MDOF systems

Abstract: This paper presents a test system for conducting on-line tests in a real time and a series of real-time on-line tests conducted to verify the effectiveness of the system. The proposed system is characterized by (1) use of a Digital Signal Processor (DSP) now readily available, (2) adoption of the C language to ensure easy programming, and (3) separation of response analysis and displacement signal generation to apply the system for tests with complex structures. To create displacement signals successively without being interrupted by the computation of equations of motion, extrapolation and interpolation procedures using present and past target displacements are developed. Base-isolated building models were chosen for the real-time on-line test. The effectiveness of the extrapolation and interpolation procedures was demonstrated through a series of real-time on-line tests applied to the models treated as SDOF structures. A five-storey base-isolated building model (treated as a six DOF structure) was tested for various ground motions, and it was verified that the system is able to simulate earthquake responses involving large displacements and large velocities. The number of DOFs that can be handled in the proposed system was investigated, and it was found that the system is capable of performing the test with reasonable accuracy for up to 10 DOF structures with a range of response frequency not greater than 3·0 Hz, or 12 DOF structures with a range of response frequency not greater than 2·0 Hz. Copyright © 1999 John Wiley & Sons, Ltd.

A novel adaptive mismatch cancellation system for quadrature IF radio receivers

Abstract: This paper investigates and resolves in-phase/quadrature phase (I/Q) imbalances between the input paths of quadrature IF receivers. These mismatches along the paths result in the image interference aliasing into the desired signal band, thus reducing the dynamic range and degrading the performance of the receivers. I/Q errors occur because of gain and phase imbalances between quadrature mixers. They are also caused by capacitor mismatches in analog-to-digital converters (A/Ds), which are designed to be identical for each input path. This paper presents a novel and feasible digital signal processing (DSP) solution for the I/Q mismatch problems. The system includes a novel complex least mean square algorithm and a modified adaptive noise canceler (signal separator) to separate the desired signal and the image noise caused by the mismatch. The noise canceler can also solve the signal leakage problem, which is that the noise reference includes signal components. This system was implemented in a Xilinx FPGA and an Analog Devices DSP chip. It was tested with a complex intermediate frequency receiver, which includes an analog front end and a complex sigma-delta modulator. Both simulation results and test results show a dramatic attenuation of the image noise. Extending applications of the system to N-path systems further indicates the robustness and feasibility of this novel adaptive mismatch cancellation system.

Energy-efficient signal processing via algorithmic noise-tolerance

Abstract: In this paper, we propose a framework for low-energy digital signal processing (DSP) where the supply voltage is scaled beyond the critical voltage required to match the critical path delay to the throughput. This deliberate introduction of input-dependent errors leads to degradation in the algorithmic performance, which is compensated for via algorithmic noise-tolerance (ANT) schemes. The resulting setup comprised of the DSP architecture operating at sub-critical voltage and the error control scheme is referred to as soft DSP. It is shown that technology scaling renders the proposed scheme more effective as the delay penalty suffered due to voltage scaling reduces due to short channel effects. The effectiveness of the proposed scheme is also enhanced when arithmetic units with a higher "delay-imbalance" are employed. A prediction based error-control scheme is proposed to enhance the performance of the filtering algorithm in presence of errors due to soft computations. For a frequency selective filter, it is shown that the proposed scheme provides 60%-81% reduction in energy dissipation for filter bandwidths up to 0.5 /spl pi/ (where 2 /spl pi/ corresponds to the sampling frequency f/sub s/) over that achieved via conventional voltage scaling, with a maximum of 0.5 dB degradation in the output signal-to-noise ratio (SNR/sub o/). It is also shown that the proposed algorithmic noise-tolerance schemes can be used to improve the performance of DSP algorithms in presence of bit-error rates of up to 10/sup -3/ due to deep submicron (DSM) noise.

Method for signal controlled switching between different audio coding schemes

Abstract: A method for signal controlled switching between audio coding schemes includes receiving input audio signals, classifying a first set of the input audio signals as speech or non-speech signals, coding the speech signals using a time domain coding scheme, and coding the nonspeech signals using a transform coding scheme. A multicode coder has an audio signal input and a switch for receiving the audio signal inputs, the switch having a time domain encoder, a transform encoder, and a signal classifier for classifying the audio signals generally as speech or non-speech, the signal classifier directing speech audio signals to the time domain encoder and non-speech audio signals to the transform encoder. A multicode decoder is also provided.

Reconfigurable pipelined 2-D convolvers for fast digital signal processing

Abstract: In order to make software applications simpler to write and easier to maintain, a software digital signal-processing library that performs essential signal- and image-processing functions is an important part of every digital signal processor (DSP) developer's toolset In general, such a library provides high-level interface and mechanisms, therefore, developers only need to know how to use algorithms, not the details of how they work Complex signal transformations then become function calls, eg, C-callable functions Considering the two-dimensional (2-D) convolver function as an example of great significance for DSP's, this paper proposes to replace this software function by an emulation on a field-programmable gate array (FPGA) initially configured by software programming Therefore, the exploration of the 2-D convolver's design space will provide guidelines for the development of a library of DSP-oriented hardware configurations intended to significantly speed up the performance of general DSP processors Based on the specific convolver, and considering operators supported in the library as hardware accelerators, a series of tradeoffs for efficiently exploiting the bandwidth between the general-purpose DSP and accelerators are proposed In terms of implementation, this paper explores the performance and architectural tradeoffs involved in the design of an FPGA-based 2-D convolution coprocessor for the TMS320C40 DSP microprocessor available from Texas Instruments Incorporated However, the proposed concept is not limited to a particular processor

Adaptive fuzzy systems for multichannel signal processing

Abstract: Processing multichannel signals using digital signal processing techniques has received increased attention lately due to its importance in applications such as multimedia technologies and telecommunications. The objective of this paper is twofold: 1) to introduce adaptive filtering techniques to the reader who is just beginning in this area and 2) to provide a review for the reader who may be well versed in signal processing. The perspective of the topic offered here is one that comes primarily from work done in the field of multichannel (color) image processing. Hence, many of the techniques and works cited here relate to image processing with the emphasis placed primarily on filtering algorithms based on fuzzy concepts, multidimensional scaling, and order statistics-based designs. It should be noted, however, that multichannel signal processing is a very broad field and thus contains many other approaches that have been developed from different perspectives, such as transform domain filtering, classical least-square approaches, neural networks, and stochastic methods, just to name a few. We present a general formulation based on fuzzy concepts, which allows the use of adaptive weights in the filtering structure, and we discuss different filter designs. The strong potential of fuzzy adaptive filters for multichannel signal applications, such as color image processing, is illustrated with several examples.

Wavefront coding: a modern method of achieving high-performance and/or low-cost imaging systems

Abstract: This paper gives a brief introduction into the background, application, and design of Wavefront Coding imaging systems Wavefront Coding is a general technique of using generalized aspheric optics and digital signal processing to greatly increase the performance and/or reduce the cost of imaging systems The type of aspheric optics employed results in optical imaging characteristics that are very insensitive to misfocus related aberrations A sharp and clear image is not directly produced from the optics, however, digital signal processing applied to the sampled image produces a sharp and clear final image that is also insensitive to misfocus related aberrations This paper gives an overview of Wavefront Coding and example images related to the two applications of machine vision/label reading and biometric imaging Design techniques of Wavefront Coding are unique from that of traditional imaging system design since both the optics and digital processing characteristics of the system are jointly optimized for optimum system performance

Accurate measurement of power system frequency using a digital signal processing technique

Abstract: This paper describes the design, computational aspects, and implementation of a digital signal processing technique for measuring the operating frequency of a power system. The technique provides correct and noise-free estimates for near-nominal, nominal, and off-nominal frequencies in about 25 ms, and it requires modest computations. The proposed technique is implemented using a DSP-based board and has been extensively tested using voltage signals obtained from a dynamic frequency source and from a power system. Some test results are presented in the paper.

Video/audio signal processing method and video/audio signal processing apparatus

Abstract: A metadata extraction unit has a feature point selection and motion estimation unit (62) for extracting at least one feature point representing characteristics of the video/audio signals in a compressed domain of the video/audio signals. Thus, reduction of time or cost for processing can be realized and it makes it possible to process effectively.

An improved weighted least-squares design for variable fractional delay FIR filters

Abstract: Digital filters capable of changing their frequency response characteristics are often referred to as variable digital filters (VDFs) and have been found useful in a number of digital signal processing applications. An important class of VDFs is the class of digital filters with variable fractional delay. This paper describes an enhanced weighted least-squares design for variable-fractional-delay finite-impulse response filters, which offers improved performance of the filters obtained with considerably reduced computational complexity compared to a recently proposed weighted least-squares (WLS) design method. The design enhancement is achieved by deriving a closed-form formula for evaluating the WLS objective function. The formula facilitates accurate and efficient function evaluations as compared to summing up a large number of discrete terms, which would be time consuming and inevitably introduce additional errors into the design.

Design of low-error fixed-width multipliers for DSP applications

Abstract: In this work, two designs of low-error fixed-width sign-magnitude parallel multipliers and two's-complement parallel multipliers for digital signal processing applications are presented. Given two n-bit inputs, the fixed-width multipliers generate n-bit (instead of 2 n-bit) products with low product error, but use only about half the area and less delay when compared with a standard parallel multiplier. In them, cost-effective carry-generating circuits are designed, respectively, to make the products generated more accurately and quickly. Applying the same approach, a low error reduced-width multiplier with output bit-width between n- and 2n has also been designed. Experimental results show that the proposed fixed-width and reduced-width multipliers have lower error than all other fixed-width multipliers and are still cost effective. Due to these properties, they are very suitable for use in many multimedia and digital signal processing applications such as digital filtering, arithmetic coding, wavelet transformation, echo cancellation, etc.

Pixel-level processing: why, what, and how?

Abstract: Pixel level processing promises many significant advantages including high SNR, low power, and the ability to adapt image capture and processing to different environments by processing signals during integration. However, the severe limitation on pixel size has precluded its mainstream use. In this paper we argue that CMOS technology scaling will make pixel level processing increasingly popular. Since pixel size is limited primarily by optical and light collection considerations, as CMOS technology scales, an increasing number of transistors can be integrated at the pixel. We first demonstrate that our argument is supported by the evolution of CMOS image sensor from PPS to APS. We then briefly survey existing work on analog pixel level processing an d pixel level ADC. We classify analog processing into intrapixel and interpixel. Intrapixel processing is mainly used to improve sensor performance, while interpixel processing is used to perform early vision processing. We briefly describe the operation and architecture of our recently developed pixel level MCBS ADC. Finally we discuss future directions in pixel level processing. We argue that interpixel analog processing is not likely to become mainstream even for computational sensors due to the poor scaling popular since it minimizes analog processing, and requires only simple and imprecise circuits to implement. We then discuss the inclusion of digital memory and interpixel digital processing in future technologies to implement programmable digital pixel sensors.

Virtual radios

Abstract: Conventional software radios take advantage of vastly improved analog to digital converters (ADCs) and digital signal processing (DSP) hardware. Our approach, which we refer to as virtual radios, also depends upon high performance ADCs. However, rather than use DSPs, we have chosen to ride the curve of rapidly improving workstation hardware. We use wideband digitization and then perform all of the digital signal processing in user space on a general purpose workstation. This approach allows us to experiment with new approaches to signal processing that exploit the hardware and software resources of the workstation. Furthermore, it allows us to experiment with different ways of structuring systems in which the radio component of communication devices is integrated with higher-level applications. This paper describes the design and performance of an environment we have constructed that facilitates building virtual radios and of two applications built using that environment. The environment consists of an input/output (I/O) subsystem that provides high bandwidth low latency user-level access to digitized signals and a programming environment that provides an infrastructure for building applications. The applications, which exemplify some of the benefits of virtual radios, are a software cellular receiver and a novel wireless network interface.

Methods and apparatus for audio-visual speaker recognition and utterance verification

Abstract: Methods and apparatus for performing speaker recognition comprise processing a video signal associated with an arbitrary content video source and processing an audio signal associated with the video signal. Then, an identification and/or verification decision is made based on the processed audio signal and the processed video signal. Various decision making embodiments may be employed including, but not limited to, a score combination approach, a feature combination approach, and a re-scoring approach. In another aspect of the invention, a method of verifying a speech utterance comprises processing a video signal associated with a video source and processing an audio signal associated with the video signal. Then, the processed audio signal is compared with the processed video signal to determine a level of correlation between the signals. This is referred to as unsupervised utterance verification. In a supervised utterance verification embodiment, the processed video signal is compared with a script representing an audio signal associated with the video signal to determine a level of correlation between the signals.

A methodology and design environment for DSP ASIC fixed point refinement

Abstract: Complex signal processing algorithms are specified in floating point precision. When their hardware implementation requires fixed point precision, type refinement is needed. The paper presents a methodology and design environment for this quantization process. The method uses independent strategies for fixing MSB and LSB weights of fixed point signals. It enables short design cycles by combining the strengths of both analytical and simulation based methods.

Analog and digital signal processing

Abstract: LIST OF TABLES / PREFACE / FROM THE PREFACE TO THE FIRST EDITION 1. OVERVIEW Introduction / Signals / Systems / The Frequency Domain / From Concept to Application 2. ANALOG SIGNALS Scope and Objectives / Signals / Operations on Signals / Signal Symmetry / Harmonic Signals and Sinusoids / Signal Symmetry / Harmonic Signals and Sinusoids / Commonly Encountered Signals / The Impulse Function / The Doublet / Moments / Problems 3. DISCRETE SIGNALS Scope and Objectives / Discrete Signals / Operations on Discrete Signals / Decimation and Interpolation / Common Discrete Signals / Discrete-Time Harmonics and Sinusoids / Aliasing and the Sampling Theorem / Random Signals / Problems 4. ANALOG SYSTEMS Scope and Objectives / Introduction / System Classification / Analysis of LTI Systems / LTI Systems Described by Differential Equations / The Impulse Response of LTI Systems / System Stability / Application-Oriented Examples / Problems 5. DISCRETE-TIME SYSTEMS Scope and Objectives / Discrete-Time Operators / System Classification / Digital Filters / Digital Filters Described by Difference Equations / Impulse Response of Digital Filters / Stability of Discrete-Time LTI Systems / Connections: System Representation in Various Forms / Application-Oriented Examples / Problems 6. CONTINUOUS CONVOLUTION Scope and Objectives / Introduction / Convolution of Some Common Signals / Some Properties of Convolution / Convolution by Ranges (Graphical Convolution) / Stability and Causality / The Response to Periodic Inputs / Periodic Convolution / Connections: Convolution and Transform Methods / Convolution Properties Based on Moments / Correlations / Problems 7. DISCRETE CONVOLUTION Scope and Objectives / Discrete Convolution / Convolution Properties / Convolution of Finite Sequences / Stability and Causality of LTI Systems / System Response to Periodic Inputs / Periodic Convolution / Connections: Discrete Convolution and Transform Methods / Deconvolution / Discrete Correlation / Problems 8. FOURIER SERIES Scope and Objectives / Fourier Series: A First Look / Simplifications Through Signal Symmetry / Parseval"s Relation and the Power in Periodic Signals / The Spectrum of Periodic Signals / Properties of Fourier Series / Signal Reconstruction and the Gibbs Effect / System Response to Periodic Inputs / Application-Oriented Examples / The Dirichlet Kernel and the Gibbs Effect / The Fourier Series, Orthogonality, and Least Squares / Existence, Convergence, and Uniqueness / A Historical Perspective / Problems 9. THE FOURIER TRANSFORM Scope and Objectives / Introduction / Fourier Transform Pairs and Properties / System Analysis Using the Fourier Transform / Frequency Response of Filters / Energy and Power Spectral Density / Time-Bandwidth Measures / Problems 10. MODULATION Scope and Objectives / Amplitude Modulation / Single-Sideband AM / Angle Modulation / Wideband Angle Modulation / Demodulation of FM Signals / The Hilbert Transform / Problems 11. THE LAPLACE TRANSFORM Scope and Objectives / The Laplace Transform / Properties of the Laplace Transform / Poles and Zeros of the Transfer Function / The Inverse Laplace Transform / The s-plane and BIBO Stability / The Laplace Transform and System Analysis / Connections / Problems 12. APPLICATIONS OF THE LAPLACE TRANSFORM Scope and Objectives / Frequency Response / Minimum-Phase Filters / Bode Plots / Performance Measures / Feedback / Application of Feedback: The Phase-Locked Loop Problems 13. ANALOG FILTERS Scope and Objectives / Introduction / The Design Process / The Butterworth Filter / The Chebyshev Approximation / The Inverse Chebyshev Approximation / The Elliptic Approximation / The Bessel Approximation / Problems 14. SAMPLING AND QUANTIZATION Scope and Objectives / Ideal Sampling / Sampling, Interpolation, and Signal Recovery / Quantization / Digital Processing of Analog Signals / Compact Disc Digital Audio / Dynamic Range Processors / Problems 15. THE DISCRETE-TIME FOURIER TRANSFORM Scope and Objectives / The Discrete-Time Fourier Transform / Connections: The DTFT and the Fourier Transform / Properties of the DTFT / The Transfer Functions / System Analysis Using the DTFT / Connections / Ideal Filters / Some Traditional and Non-traditional Filters / Frequency Response of Discrete Algorithms / Oversampling and Sampling Rate Conversion / Problems 16. THE DFT AND FFT Scope and Objectives / Introduction / Properties of the DFT / Connections / Approximating the DTFT by the DFT / The DFT of Periodic Signals / Spectral Smoothing by Time Windows / Applications in Signal Processing / Spectrum Estimation / Matrix Formulation of the DFT and IDFT / The FFT / Why Equal Lengths for the DFT and IDFT? / Problems 17. THE Z-TRANSFORM Scope and Objectives / The Two-Sided z-Transform / Properties of the z-Transform / Poles, Zeros, and the z-Plane / The Transfer Function / The Inverse z-Transform / The One-Sided z-Transform / The z-Transform and System Analysis / Frequency Response / Connections / Problems 18. APPLICATIONS OF THE Z-TRANSFORM Scope and Objectives / Transfer Function Realization / Interconnected Systems / Minimum-Phase Systems / The Frequency Response: A Graphical Interpretation / Application-Oriented Examples / Allpass Filters / Application-Oriented Examples: Digital Audio Effects / Problems 19. IIR DIGITAL FILTERS Scope and Objectives / Introduction / IIR Filter Design / Response and Matching / The Matched z-Transform for Factored Forms / Mappings from Discrete Algorithms / The Bilinear Transformation / Spectral Transformations for IIR Filters / Design Recipe for IIR Filters / Problems 20. FIR DIGITAL FILTERS Scope and Objectives / Symmetric Sequences and Linear Phase / Window-Based Design / Half-Band FIR Filters / FIR Filter Design by Frequency Sampling / Design of Optimal Linear-Phase FIR Filters / Application: Multistage Interpolation and Decimation / Maximally Flat FIR Filters / FIR Differentiators and Hilbert Transformers / Least Squares and Adaptive Signal Processing / Problems 21. MATLAB(R) EXAMPLES Introduction / ADSP Toolbox and Its Installation / MATLAB Tips and Pointers / Graphical User Interface Programs / The ADSP Toolbox / Examples of MATLAB Code / REFERENCES / INDEX

A DSP-based DS-CDMA multiuser receiver employing partial parallel interference cancellation

Abstract: The implementation of advanced DS-CDMA receivers based on multiuser detection principles is becoming a reality thanks to the combination of an improved understanding of the theoretical basis of multiuser detection and advances in digital, mixed-signal, and RF technologies. Due to their lower complexity, subtractive interference cancellation approaches are attractive for the practical implementation of multiuser detection. In a parallel interference cancellation receiver, it is practical to use the soft outputs of a matched filter bank for amplitude estimation. A bias arises in the decision statistics, however, due to imperfect estimation and interference cancellation. In this paper, the source of the bias is explicitly recognized, and a partial interference cancellation scheme that mitigates the negative effects of biased estimation and significantly improves system performance is proposed. A practical real-time algorithm that significantly reduces the implementation complexity of this scheme without sacrificing performance is then derived. To facilitate a software radio implementation, the signal processing complexity of the approach is characterized. The real-time processing algorithm is tested via implementation in software on a floating-point general-purpose DSP. The prototype includes a flexible software-based architecture which performs IF sampling and uses digital downconversion prior to baseband processing. The hardware test setup is described, and the results are presented and compared with simulation and analytical results. The experimental results confirm the simulation and analytical results which show large performance gains over the conventional matched filter.

Digital signal propagation in dispersive media.

Abstract: In this article, the propagation of digital and analog signals through media which, in general, are both dissipative and dispersive is modeled using the one-dimensional telegraph equation. Input signals are represented using impulsive, Heaviside unit step, Gaussian, rectangular pulse, and both unmodulated and modulated sinusoidal pulse type boundary data. Applications to coaxial transmission lines and freshwater signal propagation, for both digital and analog signals, are included. The analysis presented here supports the finding that digital transmission in dispersive media is generally superior to that of analog. The boundary data (input signals) give rise to solutions of the telegraph equation which contain propagating discontinuities. It is shown that the magnitudes of these discontinuities, as a function of distance, can be found without the need of solving the governing equation. Thus, for digital signals in particular, signal strength at a given distance from the input source can be easily determined. Furthermore, the magnitudes of these discontinuities are found to be independent of both the dispersion coefficient k and the elastic coefficient b. In addition, it is shown that, depending on the algebraic sign of k, one of two distinct forms of dispersion is possible and that for small-time intervals, solutions are approximately independent of k.

Theory and applications of the double-base number system

Abstract: In this paper, we analyze some of the main properties of a double base number system, using bases 2 and 3; in particular, we emphasize the sparseness of the representation. A simple geometric interpretation allows an efficient implementation of the basic arithmetic operations and we introduce an index calculus for logarithmic-like arithmetic with considerable hardware reductions in lookup table size. We discuss the application of this number system in the area of digital signal processing; we illustrate the discussion with examples of finite impulse response filtering.

Experimental ultrasound system for real-time synthetic imaging

Abstract: Digital signal processing is being employed more and more in modern ultrasound scanners. This has made it possible to do dynamic receive focusing for each sample and implement other advanced imaging methods. The processing, however, has to be very fast and cost-effective at the same time. Dedicated chips are used in order to do real time processing. This often makes it difficult to implement radically different imaging strategies on one platform and makes the scanners less accessible for research purposes. Here flexibility is the prime concern, and the storage of data from all transducer elements over 5 to 10 seconds is needed to perform clinical evaluation of synthetic and 3D imaging. This paper describes a real-time system specifically designed for research purposes. The purpose of the system is to make it possible to acquire multi-channel data in real-time from clinical multi-element ultrasound transducers, and to enable real-time or near real-time processing of the acquired data. The system will be capable of performing the processing for the currently available imaging methods, and will make it possible to perform initial trials in a clinical environment with new imaging modalities for synthetic aperture imaging, 2D and 3D B-mode and velocity imaging. The system can be used with 128 element transducers and can excite 128 channels and receive and sample data from 64 channels simultaneously at 40 MHz with 12 bits precision. Data can be processed in real time using the system's 80 signal processing units or it can be stored directly in RAM. The system has 24 GBytes RAM and can thus store 8 seconds of multi-channel data. It is fully software programmable and its signal processing units can also be reconfigured under software control. The control of the system is done over an Ethernet using C and Matlab. Programs for doing, e.g. B-mode imaging can directly be written in Matlab and executed on the system over the net from any workstation running Matlab. The overall system concept is presented and an example of a 20 lines script for doing phased array B-mode imaging is presented.

Architecture for a signal collection system of a wireless location system

Abstract: A signal collection system (SCS) for use in a Wireless Location System is disclosed. The SCS performs wideband energy detection and reporting at the front end of the SCS receiver. Other aspects of the SCS include a protocol for efficiently setting levels for wideband energy detection, DSP sharing within an SCS, and recursive location processing using progressively greater bandwidth from temporarily stored wideband data. The disclosed SCS includes antennas, a wideband receiver, a DSP for wideband energy detection, a memory for temporarily storing digital samples of received signals, a digital drop receiver, demodulation and normalization processors, and a communications processor. The wideband energy detection and the demodulation and normalization processors are implemented with DSP's that detect energy in a particular band, demodulate selected signals, and extract signals of interest for forwarding. The wideband energy detection unit determines the presence of a transmitted signal in the control channels monitored by the SCS. The wideband energy detection involves forming a map of the channel spectrum, and the map is used to determine when to demodulate signals within selected channels.

Binaural-signal-processing techniques

Abstract: A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined fron the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

Automatic loudspeaker equalizer

Abstract: A method for generating digital filters for equalizing a loudspeaker. First digital data is provided, for a tolerance range for a target response curve of sound level versus frequency for the loudspeaker. Second digital data is generated, for an actual response curve of sound level versus frequency for the loudspeaker (1010). The first digital data is compared with the second digital data and it is determined whether the actual response curve is within the tolerance range (1020). If the actual response curve is not within the tolerance range, digital audio filters are iteratively generated, and the digital audio filters are applied to the second digital data to generate third digital data for a compensated response curve (1050, 1060, 1070). The frequency, amplitude and bandwidth of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range or a predetermined limit on the number of digital audio filters has been reached, whichever occurs first (1080).