Showing papers on "Digital signal processing published in 1996"

Statistical Digital Signal Processing and Modeling

Abstract: From the Publisher: The main thrust is to provide students with a solid understanding of a number of important and related advanced topics in digital signal processing such as Wiener filters, power spectrum estimation, signal modeling and adaptive filtering. Scores of worked examples illustrate fine points, compare techniques and algorithms and facilitate comprehension of fundamental concepts. Also features an abundance of interesting and challenging problems at the end of every chapter.

Active Noise Control Systems: Algorithms and DSP Implementations

Abstract: From the Publisher: Active noise control (ANC) is rapidly becoming the most effective way to reduce noises that can otherwise be very difficult and expensive to control ANC is achieved by introducing a canceling "anti-noise" wave through an appropriate array of secondary sources When applied accurately, ANC can provide effective solutions to noise-related problems in a broad range of areas, including manufacturing and industrial operations as well as consumer products Consequently, ANC research and development has become an important focus of both industrial applications and engineering research Active Noise Control Systems: Algorithms and DSP Implementations introduces the basic concepts of ANC with an emphasis on digital signal processing (DSP) hardware and adaptive signal processing algorithms, both of which have come into prominence within the last decade The authors emphasize the practical aspects of ANC systems by combining the principles of adaptive signal processing with both experimental results and practical implementation Applications are cited in many fields and encompass all types of noise media, including air-acoustic, hydroacoustic, vibrations, and others The specific implementation stressed is based on the TMS320 family of signal processors from Texas Instruments, which are the most widely used worldwide Coverage of theory includes concise derivations and analyses of commonly used adaptive structures and algorithms for active noise control applications, which are enhanced by the inclusion of a floppy disk featuring C and assembly programs for implementing many ANC systems Mathematical representations are employed and the source code included on the disk is in a form that is easily accessible to anyone using the book For practicing engineers, researchers, and advanced students in signal processing, Active Noise Control Systems: Algorithms and DSP Implementations will serve as a comprehensive, state-of-the-art text/reference on this important and rapidly de

GPS receiver and method for processing GPS signals

Abstract: A GPS receiver in one embodiment includes an antenna which receives GPS signals at an RF frequency from in view satellites; a downconverter coupled to the antenna for reducing the RF frequency of the received GPS signals to an intermediate frequency (IF); a digitizer coupled to the downconverter and sampling the IF GPS signals at a predetermined rate to produce sampled IF GPS signals; a memory coupled to the digitizer storing the sampled IF GPS signals (a snapshot of GPS signals); and a digital signal processor (DSP) coupled to the memory and operating under stored instructions thereby performing Fast Fourier Transform (FFT) operations on the sampled IF GPS signals to provide pseudorange information. These operations typically also include preprocessing and post processing of the GPS signals. After a snapshot of data is taken, the receiver front end is powered down. The GPS receiver in one embodiment also includes other power management features and includes, in another embodiment the capability to correct for errors in its local oscillator which is used to sample the GPS signals. The calculation speed of pseudoranges, and sensitivity of operation, is enhanced by the transmission of the Doppler frequency shifts of in view satellites to the receiver from an external source, such as a basestation in one embodiment of the invention.

A course in digital signal processing

Abstract: From the Publisher: This book provides readers with a precise, comprehensive, practical, and up-to-date exposition on digital signal processing. Both mathematical and useful, it presents a rigorous course of study to help readers learn the theory and practice of DSP. Porat includes physical and engineering application, coupled with mathematical derivations to the extent necessary for understanding DSP concepts and methods. The book contains detailed discussion of practical spectral analysis, including the use of windows for spectral analysis, sinusoidal signal analysis, and the effect of noise. There is also comprehensive treatment of both FIR and IIR filters, including detailed design procedures and MATLAB tools.

Signal processing with fractals: a wavelet-based approach

Abstract: Wavelet transmission statistically self-similar signals detection and estimation with 1/processes deterministically self-similar signals fractal modulation linear self-similar signals.

Wavelet analysis [for signal processing]

Abstract: As every engineering student knows, any signal can be portrayed as an overlay of sinusoidal waveforms of assorted frequencies. But while classical analysis copes superbly with naturally occurring sinusoidal behavior-the kind seen in speech signals-it is ill-suited to representing signals with discontinuities, such as the edges of features in images. Latterly, another powerful concept has swept applied mathematics and engineering research: wavelet analysis. In contrast to a Fourier sinusoid, which oscillates forever, a wavelet is localized in time-it lasts for only a few cycles. Like Fourier analysis, however, wavelet analysis uses an algorithm to decompose a signal into simpler elements. Here, the authors describe how localized waveforms are powerful building blocks for signal analysis and rapid prototyping-and how they are now available in software toolkits.

Differential evolution design of an IIR-filter

Abstract: The task of designing an 18 parameter IIR-filter (IIR=infinite impulse response) which has to meet tight specifications for both magnitude response and group delay is investigated. This problem is usually tackled by specialized design methods and requires an expert in digital signal processing for its solution. The use of the general purpose minimization method differential evolution (DE), however, allows filter design with a minimum knowledge of digital filters.

Ultrasonic diagnostic imaging system with personal computer architecture

Abstract: An ultrasonic diagnostic imaging system is provided with a personal computer platform which processes digital echo signals and produces ultrasonic image signals for display. The expansion bus structure of the personal computer platform accommodates ancillary processors such as beamformer cards, digital signal processing cards, video cards, and network cards which may be necessary or desirable for the ultrasound system. In a preferred embodiment the digital signal samples produced by a beamformer connected to the expansion bus are processed for display by software executed by the CPU of the personal computer platform. A preferred software architecture for the personal computer based ultrasound system consists of multiple object oriented software tasks, executing under a realtime, multitasking operating system which is both efficient and robust. Performance upgrades of the entire ultrasound system are effected by simple replacement of the CPU with a higher performance CPU, thus providing continual ultrasound system performance improvements in consonance with the evolution of personal computer CPU technology.

Speech signal transmitting and receiving apparatus with noise sensitive volume control

Abstract: A speech signal transmitting receiving apparatus, such as a portable telephone set, includes a speech signal transmitting encoding circuit, a noise domain detection unit, a noise level detection unit and a controller. The speech signal transmitting encoding circuit compresses input speech signals by digital signal processing at a high efficiency. The noise domain detection unit detects the noise domain using an analytic pattern produced by the speech signal transmitting encoding circuit. The noise level detection unit detects the noise level of the noise domain detected by the noise domain detection unit. The controller controls the received sound volume responsive to the noise level detected by the noise level detection unit.

Common receive module for a programmable digital radio

Abstract: A digital submodule is included in a software programmable common receive module for receiving intermediate frequency signals and producing a serial bit stream. The digital submodule is programmable based on a selected application of a plurality of radio applications and, if present, a selected function of a plurality of functions of the selected radio application. The digital submodule may include an analog to digital converter for converting intermediate frequency signals received from an analog submodule into digital signals. The digital signals are supplied to a programmable signal processing unit which is configured, according to the selected radio application and, if present, the selected function, to perform control functions, processing and analysis of the digital signals and generate output signals. The output signals are then formatted by a formatting unit producing formatted digital signals. The formatted digital signals are then supplied to a system bus. The programmable signal processing unit may include a digital downconverter for selective use depending on the selected application of radio communication, for generating a baseband signal. Additionally, a central processing unit is included to perform further signal processing for selected radio applications.

Area-efficient multipliers for digital signal processing applications

Abstract: An area-efficient parallel sign-magnitude multiplier that receives two N-bit numbers and produces an N-bit product, referred to as a truncated multiplier, is described. The quantization of the product to N bits is achieved by omitting about half the adder cells needed to add the partial products but in order to keep the quantization error to a minimum, probabilistic biases are obtained and are then fed to the inputs of the retained adder cells. The truncated multiplier requires approximately 50% of the area of a standard parallel multiplier. The paper then shows that this design strategy can also be applied for the design of two's-complement multipliers. The paper concludes with the application of the truncated multiplier for the implementation of a digital filter and it is shown that the signal-to-noise ratio of the digital filter using a truncated multiplier is better than that using a standard multiplier.

GPS receiver having power management

Abstract: A GPS receiver having a low power mode of operation in one embodiment includes an antenna which receives GPS signals at an RF frequency from in view satellites; a downconverter coupled to the antenna for reducing the RF frequency of the received GPS signals to an intermediate frequency (IF); a digitizer coupled to the downconverter and sampling the IF GPS signals at a predetermined rate to produce sampled IF GPS signals; a memory coupled to the digitizer storing the sampled IF GPS signals; and a digital signal processor (DSP) coupled to the memory and operating under stored instructions thereby performing operations on the sampled IF GPS signals to provide pseudorange information. In one example, after the sampled IF GPS signals have been stored in the memory, the GPS receiver front end is powered down and the DSP is powered up. The GPS receiver in one embodiment also includes other power management features.

Digital signal processing for linearization of small input signals to a tri-state power switch

Abstract: An all digital switching amplifier includes an input over-sampling filter (20) for receiving a pulse code modulated (PCM) digital input signal. Oversampled PCM data are supplied to a multibit noise shaper (22), which frequency shapes quantization error. The oversampled, noise-shaped, PCM data is applied to an amplitude-to-time converter (24), which produces variable-width command pulses. The command pulses from converter (24) are applied to switch drive logic circuit (28) to enable switches (26) to connect a filter (30) and load (32) to power supply (34).

Low-power digital filtering using approximate processing

Abstract: We present an algorithmic approach to the design of low-power frequency-selective digital filters based on the concepts of adaptive filtering and approximate processing. The proposed approach uses a feedback mechanism in conjunction with well-known implementation structures for finite impulse response (FIR) and infinite impulse response (IIR) digital filters. Our algorithm is designed to reduce the total switched capacitance by dynamically varying the filter order based on signal statistics. A factor of 10 reduction in power consumption over fixed-order filters is demonstrated for the filtering of speech signals.

Data driven signal processing: an approach for energy efficient computing

Abstract: The computational switching activity of digital CMOS circuits can be dynamically minimized by designing algorithms that exploit signal statistics. This results in processors that have time-varying power requirements and perform computation on demand. An approach is presented to minimize the energy dissipation per data sample in variable-load DSP systems by adaptively minimizing the power supply voltage for each sample using a variable switching speed processor. In general, using buffering and filtering, the computation can be spread over multiple samples averaging the workload and lowering energy further. It is also shown that four levels of voltage quantization combined with dithering is sufficient to closely emulate arbitrary voltage levels.

Accurate distance measurement by an autonomous ultrasonic system combining time-of-flight and phase-shift methods

Abstract: Ultrasonic sensor distance measurements are based on the evaluation of the time-of-flight or on the determination of the phase difference between a transmitted and a received signal. Digital signal processing methods are developed in order to extract the useful information from the samples acquired from the received wave. This paper presents an efficient algorithm for distance measurement combining both the propagation time method and the phase difference method and relying on a low rate sampling technique allowed by the limited bandwidth of the ultrasonic transducers. The measuring system was implemented and tested on a compact Motorola MC68HC16 based platform, with a minimum of attached hardware. Experimental results show an accuracy better than 1 mm.

Optimal PWM based on real-time solution of harmonic elimination equations

Abstract: Pulse-width modulation of DC/AC power converters (PWM) based on the elimination of low-order harmonics necessitates solving systems of nonlinear equations. Conventional implementations of this technique based on storing off-line calculated solutions have the common problem that the system flexibility is very limited, especially for applications that require both amplitude and frequency control. A new implementation scheme based on real-time solution of nonlinear harmonic elimination equations using a digital signal processor DSP56001 is reported in this paper. With this digital signal processor (DSP), optimal pulse patterns having 15 switching angles in each quarter fundamental period can be determined within 2.15 ms. Details of the system hardware and software are described. New theoretical results concerning the solvability of harmonic elimination equations are also presented.

User independent, real-time speech recognition system and method

Abstract: A system and method for identifying the phoneme sound types that are contained within an audio speech signal is disclosed. The system includes a microphone (12) and associated conditioning circuitry (14), for receiving an audio speech signal and converting it to a representative electrical signal. The electrical signal is then sampled and converted to a digital audio signal with a digital-to-analog converter (34). The digital audio signal is input to a programmable digital sound processor (18), which digitally processes the sound so as to extract various time domain and frequency domain sound characteristics. These characteristics are input to a programmable host sound processor (20) which compares the sound characteristics to standard sound data. Based on this comparison, the host sound processor (20) identifies the specific phoneme sounds that are contained within the audio speech signal. The programmable host sound processor (20) further includes linguistic processing program methods to convert the phoneme sounds into English words or other natural language words. These words are input to a host processor (22), which then utilizes the words as either data or commands.

Compensation method for I/Q channel imbalance errors

Abstract: A method in which correctable I/Q imbalance errors in a complex receiver can be detected and compensated for digitally without the use of special calibration signals. Differential D.C. offset errors are compensated by averaging the incoming I d and Q d digital signals and subtracting from them an expected value of differential D.C. offset, for example, computed from the long term average of the I and Q signals to create I' and Q' signals. Differential gain imbalance errors are corrected by calculating a root means square average of the I' and Q' digital signals and applies to them compensation coefficients K x and K y determined from either the RMS average or from a Stochastic Gradient Algorithm. The DSP compensates for the quadrature phase errors by calculating a compensation matrix which is independent of the frequency of the carrier and applies the compensation matrix to the I' and Q' digital signals. The compensation matrix for quadrature phase errors is completely independent of the frequency of the input carrier signal supplied to the complex receiver, and is not dependent on the use of a calibration signal. The compensation may be performed as a step in calibration of the complex receiver, or continuously.

An experimental comparison of robust control algorithms on a direct drive manipulator

Abstract: In this paper we present an experimental comparison of recent passivity based robust control algorithms on a two-link direct-drive robot arm. The manipulator is actuated with high-torque brushless DC motors and is controlled by a digital signal processor (DSP) development system interfaced to a PC486 workstation. Four algorithms are compared with respect to ease of design, implementation, and performance of the closed-loop systems.

Process control transmitter

Abstract: A transmitter (10) in a process control system includes input/output circuitry (74) for coupling to a process control loop (12). A first sensor (40L) having a first impedance is responsive to a first sensed parameter. A second sensor (40H) having a second impedance is responsive to a sensed parameter. First and second excitation signals (S1, S2) are applied to the first and second sensors (40L, 40H). A summing node (44) sums the outputs (OL, OH) of the first and second sensors (40L, 40H). An analog to digital converter (54) provides a digital output representative of the summed signals. Digital signal processing circuitry (70) coupled to the analog to digital converter (54) provides an output related to the outputs (OL, OH) of the first and second sensors (40L, 40H) to the input/output circuitry (72) for transmission over the process control loop (12).

Low bit rate high quality audio coding with combined harmonic and wavelet representations

Abstract: We describe a novel high quality audio coding method using adaptive signal representation, based on sinusoidal and wavelet analysis of signals. First, we perform a harmonic analysis of the signal to remove strong periodic structures or tones from the signal. Then we carry out wavelet analysis that are useful in tracking the transients of the signal. These transients are then removed from the wavelet coefficients. The remaining coefficients have broadband noise-like structure. Since this method separates out tones (sinusoids), transients, and broadband noise, we may use tonal, noise, and temporal masking information to individually encode the tones and the wavelet coefficients. Our experiments suggest that this method yields a nominal bit rate of 1 bit/sample for high quality audio compression.

Method and apparatus for adaptive line enhancement in coriolis mass flow meter measurement

Abstract: An apparatus and method for determining frequency and phase relationships of vibrating flow tubes in a Coriolis mass flow meter. Adaptive line enhancement (ALE) techniques and apparatus are used in a digital signal processing (DSP) device to accurately determine frequency and phase relationships of the vibrating flow tube and to thereby more accurately determine mass flow rate of a material flowing through the mass flow meter. In a first embodiment, an adaptive notch filter is used to enhance the signal from each corresponding sensor signal on the vibrating flow tubes. In a second embodiment, a plurality of adaptive notch filters are cascaded to enhance the signal from each corresponding sensor signal. In both embodiments, an antialiasing decimation filter associated with each sensor signal reduces the computational complexity by reducing the number of samples from a fixed frequency A/D sampling device associated with each sensor signal. Computational adjustments are performed to compensate for spectral leakage between the fixed sampling frequency and the variable fundamental frequency of the vibrating flow tubes. Despite this added computational complexity, the present invention is simpler than prior designs and provides better noise immunity due to the adaptive notch filtration. Heuristics are applied to the weight adaptation algorithms of the notch filters to improve convergence of the digital filters and to reduce the possibility of instability of the filters interfering with mass flow measurements.

Guide to using field programmable gate arrays (FPGAs) for application-specific digital signal processing performance

Abstract: FPGAs have become a competitive alternative for high performance DSP applications, previously dominated by general purpose DSP and ASIC devices. This paper describes the benefits of using an FPGA as a DSP co-processor, as well as, a stand-alone DSP engine. Two case studies, a Viterbi decoder and a 16-tap FIR filter are used to illustrate how the FPGA can radically accelerate system performance and reduce component count in a DSP application. Finally, different implementation techniques for reducing hardware requirements and increasing performance are described in detail.

Sampling in digital signal processing and control

Abstract: Fourier analysis sampling and reconstruction analysis of discrete-time systems discrete-time models of continuous deterministic systems optimal linear estimation with finite impulse response filters optimal linear estimation with state-space filters periodic and multirate filtering discrete time control sampled data control generalized sample-hold functions periodic control of linear time-invariant systems multirate control optimal control of periodic systems

Analysis of myoelectric signals recorded during dynamic contractions

Abstract: This article presents the theoretical considerations that justify the choice of specific time-frequency transforms for processing nonstationary myoelectric signals as a method of studying fatigue prior to the failure point. It shows some preliminary results obtained by applying these techniques to computer-synthesized realizations of stochastic processes, as well as to real signals detected during different types of dynamic contractions of healthy human volunteers. Five different time-frequency transforms were applied in this study (the Wigner-Ville, the smoothed Wigner-Ville, the Cone kernel, the reduced interference, and the Choi-Williams), but for the sake of brevity, this article reports only the results obtained by applying the Choi-Williams transform, because the authors found it to be the most suitable for processing these specific signals.

DSP control of high-power UPS systems feeding nonlinear loads

Abstract: Many facilities, such as patient health care centers, data processing systems, and critical telecommunication links, rely on uninterruptible power supplies (UPS) to maintain a continuous supply of power in case of a line outage. In addition to requiring continuous power, many critical nonlinear loads are sensitive to incoming line transients and input harmonic voltage distortion. Conventional UPS systems operate to protect against such disturbances using complex filtering schemes, often employing large passive components. This paper presents the advantages of using real time digital signal processing (DSP) control of UPS systems. A DSP controlled UPS inverter and harmonic conditioning system is described and the performance is verified on a 150 kVA system.

Quantization analysis and design of a digital predistortion linearizer for RF power amplifiers

Abstract: Significant improvements in terms of reduced power consumption and increased bandwidth are obtained if a digital predistortion linearizer is implemented with an application specific digital signal processor. This paper investigates the quantization effects in different parts of a table based complex gain predistortion linearizer. The analysis can be used to optimize the predistortion linearizer with respect to word length based on the knowledge of the RF amplifier gain characteristic, the probability density function for the modulation scheme and the maximum allowable adjacent channel interference level. A predistorter chip is described that has been designed using the analysis. The chip has been fabricated and tested. Compared with a standard digital signal processing (DSP) solution it provides seven times higher bandwidth but consumes only 10% of the power.

A Simplified Approach to Image Processing: Classical and Modern Techniques in C

Abstract: From the Publisher: Image processing, the use of computers to process pictures, has revolutionized the fields of medicine, space exploration, geology, and oceanography, and has become the hottest area in digital signal processing. This book provides a comprehensive introduction to the most popular image processing techniques used today, without getting bogged down in the complex mathematical presentations found in most image processing books and journals. The book covers the hottes t topics in image proessing, including whole chapters on the processing of color images, image warping and morphing techniques, and image compression. The diskette, written in portable C code, provides a "hands-on" introduction to image processing techniques that can be incorporated into the user's applications. For computer programmers and electrical engineers who need to enhance image processing applications.