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Papers by fred harris

FAME is designed to perform an all-sky, astrometric survey with unprecedented accuracy. It will c... more FAME is designed to perform an all-sky, astrometric survey with unprecedented accuracy. It will create a rigid astrometric catalog of 4 ×10 7 stars with 5 < m V < 15. For bright stars, 5 < m V < 9, FAME will determine positions and parallaxes accurate to < 50 µas, with proper motion errors < 50 µas/year. For fainter stars, 9 < m V < 15, FAME will determine positions and parallaxes accurate to < 500 µas, with proper motion errors < 500 µas/year. It will also collect photometric data on these 4 ×10 7 stars in four Sloan DSS colors.

Multirate signal processing can reduce costs and improve performance in applications ranging from... more Multirate signal processing can reduce costs and improve performance in applications ranging from laboratory instruments to cable modems, wireless systems, and consumer entertainment products. This book offers the first systematic, clear, and intuitive introduction to multirate ...

Handbook of Digital Signal Processing

Publisher Summary The discrete Fourier transform (DFT), implemented by one of the computationally... more Publisher Summary The discrete Fourier transform (DFT), implemented by one of the computationally efficient fast Fourier transform (FFT) algorithms, has become the core of many digital signal processing systems. These systems can perform general time domain signal processing and classical frequency domain processing. This chapter describes some signal processing applications that use the DFT to perform specific time domain filtering tasks. Two basic filtering tasks can be performed with DFT: (1) convolution or correlation between two arbitrary arrays and (2) narrowband channelization. The DFT gives access to the computational efficiency of the FFT. Some perspectives have evolved that allow classical filtering operations to be implemented by the FFT. The chapter highlights these perspectives. It describes the way in which a judicious choice of window, overlap, block size, and circular indexing, coupled with simple pre- and postprocessing tasks, leads to use of the DFT to perform general time domain processing. It examines the way by which simple pre- and postprocessing tasks significantly enhance the degrees of freedom available to the designer of multichannel processors, the way by which processing parameters are coupled, and the way by which a choice of processing parameters impacts important system considerations such as total computational burden and classical fidelity measures such as channel crosstalk and noise levels.

2017 22nd International Conference on Digital Signal Processing (DSP)

This paper presents a filter for exploiting timevarying cyclostationary statistics. The motivatin... more This paper presents a filter for exploiting timevarying cyclostationary statistics. The motivating example is designing redundancies into an OFDM signal to reject in-band, strong, wideband interference. Combining the filter with error correcting codes gives a 10 dB gain over LDPC codes and more than a 1000x reduction in BER. This allows the creation of multicarrier waveforms that are robust, reliable and resistant to interference. The filter improves upon existing techniques thought the ability to exploit time-varying spectral redundancy, a capability unique to this filter. The design equations for the optimal filter weights are derived, showing the upper bound for the performance of the filter. The applications range from control channel protection for next generation commercial broadband to mission-critical communications.

Fast Algorithms for Signal Processing

Conference Record of Thirty-Fifth Asilomar Conference on Signals, Systems and Computers (Cat.No.01CH37256), 2001

A digitally controlled sampled data delay line is implemented with recursive all-pass filter sect... more A digitally controlled sampled data delay line is implemented with recursive all-pass filter sections. This is in marked contrast to the standard implementation of programmable time delays that use fixed non-recursive polyphase stages or adjustable Farrow FIR filters. The recursive filter exhibits an equal-ripple approximation to constant group delay. The phase slope is programmable to present a continuously variable time delay network. Applications of a continuously adjustable, linear-time delay structure offers unique signal processing options to address various communication system tasks. These include timing recovery in DSP based receivers, adaptive beamforming and steering, communication systems channel modeling, and reverberation modeling in acoustic chambers and instruments.

Proceedings of the 5th International ICST Conference on Cognitive Radio Oriented Wireless Networks and Communications, 2010

Proceedings of 3rd International Conference on Algorithms and Architectures for Parallel Processing

The introduction of SRAM-based field programmable gate arrays (FPGAs) has opened-up a new dimensi... more The introduction of SRAM-based field programmable gate arrays (FPGAs) has opened-up a new dimension to parallel computing architectures. This paper describes an alternative approach to parallel computingreconfiguruble or virtual parallel processing (VPP). Rather than mapping an application onto a given parallel machine, the VPP approach synthesizes the appropriate type and number of processing elements, as well as the interconnection topology, that is optimal for the application. For each application, configuration data is downloaded to the machine that personalizes the hardware for the task at hand. The paper provides a brief description of the authors reconfigurable computer, Archimedes. The benefits of the VPP approach are highlighted by an example applicationthe 2-D FFT. A novel parallel implementation of a polynomial transform based 2-D transform is described and compared to results for distributed memory parallel machines that have been reported in the literature. The comparison highlights the computational advantage provided by reconfigurable computing.

Conference Record of The Twenty-Ninth Asilomar Conference on Signals, Systems and Computers

... These samples are re-stricted to the quinary set Q3 = {fl, f0.5, o}. Each sample P ~ 3 ( n ..... more ... These samples are re-stricted to the quinary set Q3 = {fl, f0.5, o}. Each sample P ~ 3 ( n ... by a factor of 1/2, version of the filter coefficient for input to the BSPC adder/subtractor unit. Half of the filter coeffi-cient value can therefore be added or subtracted from a BSPC input stream. ...

Proceedings of the First International Conference on Computer Supported Education, 2009

This paper presents MATLAB-based demo programs for transformation of normal and uniform random va... more This paper presents MATLAB-based demo programs for transformation of normal and uniform random variables. Linear as well as nonlinear transforms are considered. It is demonstrated how the transform changes the corresponding probability density function. It is also shown how to use the transformation of given random variable to generate a new desired random variable. The programs can be used as a complement to theoretical classes or alone as a self-study tool.

2019 53rd Asilomar Conference on Signals, Systems, and Computers, 2019

A Frequency Shifter shifts the full spectrum of an audio input signal up and down the frequency a... more A Frequency Shifter shifts the full spectrum of an audio input signal up and down the frequency axis. When shifted, the harmonic relationships of the input signal are not preserved and the resulting inharmonic signal has an interesting uncommon metallic sound. The frequency shifters, originally developed to supress audio feedback howling, were used in early music synthesizers to create unique special sound effects. Early frequency shifters were implemented with analogue diode ring modulators that suffered from annoying audio carrier tone bleed-through due to mixer imbalance. Implementing the frequency shifter in DSP holds the promise of reduced cost, improved dynamic range, and absence of undesired artifacts.

Proceedings of 13th International Conference on Digital Signal Processing

The Farrow filter (1988) is a multirate filter structure which offers the option of continuously ... more The Farrow filter (1988) is a multirate filter structure which offers the option of continuously adjustable resample ratio. This paper presents a derivation of the method proposed by Farrow, and demonstrates the performance and complexity of resampling filters using his technique. The paper also develops some important system options made available to the designer as spin-offs of the derivation

Conference Record of the Thirty-Eighth Asilomar Conference on Signals, Systems and Computers, 2004.

Abstract An arbitrary resampler performs interpolation of a sampled data signal, computing sample... more Abstract An arbitrary resampler performs interpolation of a sampled data signal, computing sample values of an underlying analog signal located on one set of periodically spaced sample locations from data samples located on another set of periodically spaced sample ...

2000 10th European Signal Processing Conference, 2000

Synchronization techniques based on DSP implementations are often digital emulations of their ana... more Synchronization techniques based on DSP implementations are often digital emulations of their analog prototypes. Such solutions do not include structures and algorithms responsive to DSP system considerations and implementation strengths and weaknesses. We present a number of unconventional algorithms and structures used in carrier and timing recovery schemes. Multirate signal processing, polyphase filter structures, and CORDIC subsystems are at the heart of efficient first principle DSP based solutions to carrier recovery, matched filtering, timing recovery, and phase detection tasks required for synchronization.

1996 IEEE International Symposium on Circuits and Systems. Circuits and Systems Connecting the World. ISCAS 96

This paper addresses the problem of implementing narrow-band FIR filters using FPGAs. A method ba... more This paper addresses the problem of implementing narrow-band FIR filters using FPGAs. A method based on re-quantization of the input data stream using a sigma-delta modulator is presented. The reduced bit length representation of the re-quantized input samples removes the requirement for a full multiplier in the filter hardware. The filtering technique is described and implementation results using a Xilinx XC4010 FPGA are presented. Using a bit-serial approach, a 200 tap narrow-band filter operating at a sample rate of 1.56 MHz has been developed.

Frequency hopping (FH) is a spread spectrum transmission technique that achieves frequency divers... more Frequency hopping (FH) is a spread spectrum transmission technique that achieves frequency diversity gain over frequency selective fading channels and also has a low probability of interception. This technique has been widely used in military applications, for its recognized antijamming performance, and in some wireless standards such as GSM and Bluetooth, for its interference resistance. In this paper we present a fully digital architecture for performing frequency hopped modulation. The proposed structure is composed of a cascade of two polyphase up converter channelizers. The first one performs the M-FSK modulation of the baseband signals while the second one accomplishes the task of hopping the FSK modulated spectra under the control of a pseudorandom sequence generator. According to the authors' knowledge, a fully digital architecture for frequency hopped transmission has never been presented in the literature until now. In this paper, both theoretical aspects and simulatio...

1. ABSTRACT Parametric filters use separate and non interacting controls to change bandwidth, cen... more 1. ABSTRACT Parametric filters use separate and non interacting controls to change bandwidth, center frequency, and cut or boost levels of an acoustic signal. These are a generalization of graphic equalizers which permits gain adjustment of a filter bank with fixed bandwidth and center frequencies. This paper describes and demonstrates the performance of a digital realization of a parametric filter.

In this chapter, we examine filter banks as channelizers for DSP-intensive radio transmitters and... more In this chapter, we examine filter banks as channelizers for DSP-intensive radio transmitters and receivers. On the transmitter side, a synthesis channelizer forms a composite broadband output signal from a set of narrowband baseband input signals. The analysis channelizer reverses the process in the receiver forming a set of narrowband baseband output signals from composite broadband input signals. The two types of filter banks are each other's duals. The filter banks are remarkable in their capability, flexibility, and efficiency in the tasks they perform. Central to their use is their ability to change sample rate while changing bandwidth and to move signals between different spectral regions using aliasing and to separate aliases with phase-coherent sums.

Proceedings of ICUPC - 5th International Conference on Universal Personal Communications

Gain and phase mismatch between the two signal processing paths of an I-Q receiver are responsibl... more Gain and phase mismatch between the two signal processing paths of an I-Q receiver are responsible for artifacts which limit the dynamic range of a communication system. We describe and demonstrate a technique to adaptively balance the gain and phase of the I-Q paths with DSP based balancing loops which operate as background tasks while processing input data.