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Lab VIEW Digital Signal Processing

Lab VIEW Digital Signal Processing

During the present quarter we have made some architectural changes to the five-processor ring-structured programmable digital filter IC to improve the system performance. The main change is to insert a register between the coefficient RAM and the multiplier to eliminate the read time of the RAM from the critical path of the multiplier. This provides a substantial improvement in the system performance since the critical path of the multiplier determines the performance of the over-all five-processor system. Figure 1 shows a block diagram of the new ALU architecture. A TinyChip comprising the multiplier with the redesigned carry-select vector-merge adder described in the previous report has been sent to MOSIS for fabrication. The IC consists of the 12-bit by 11-bit multiplier, the coefficient and data input registers, the output register, and RAM to store the coefficient and input data. The multiplier itself consists of 3100 transistors occupying an area of 1.53 mm2 (1.313 mm by 1.166 mm) in 2- micron CMOS technology and is simulated to operate in 22 ns. We expect the prototype parts in the middle of the next quarter for testing. Figure 2 shows the architecture of the TinyChip submitted for fabrication and Figure 3 shows the layout of this IC.

The technical objective of this project is the development and evaluation of various digital signal processing (DSP) algorithms that will enhance hybrid lidar performance. Practical algorithms must be developed taking into account the underwater propagation channel and the processing requirements for each algorithm.

The technical objective of this project is the development and evaluation of various digital signal processing (DSP) algorithms that will enhance hybrid lidar performance. Practical algorithms must be developed taking into account the underwater propagation channel and the processing requirements for each algorithm. The hardware platforms being considered for lidar signal processing are based on software defined radios (SDRs). NAWC has selected a digital hardware platform based on COMBLOCK software defined radio (SDR) modules. Clarkson has investigated several other SDR approaches which include two open source SDR platforms and one commercial platform. We selected the Signal Hound SDR for . additional characterization since it is commercially available and has a well-supported Application Programmer's Interface or API. The API allows us to configure the SDR and obtain data in a MATLAB programming environment. The primary activity during this reporting period consisted of evaluating the Signal Hound SDR for lidar signal processing. Specifically we developed software that allows us to configure and collect data from the SDR within the MATLAB programming,environment. This will allow us to used MALAB as a DSP development platform.

A signal processing software package was generated for a Data General Eclipse S/250 minicomputer. The model 4331 A/D/A converter was utilized to perform general purpose A/D/A operations and to collect, edit, and play back speech data files. The model 130 array processor was used to perform high-speed convolution and Fourier Transform related operations. The Parks-McClellan algorithm was implemented to allow design of linear phase, impulse response filters. Self-explanatory interactive programs for data collection and filter design, together with single line commands for signal processing functions, make this a simple to operate, versatile package for digital signal processing. (Author)

A class of multidimensional signals, based on what we call Generalized Group Alphabets, is introduced, and its basic properties are derived. The combination of Generalized Group Alphabets and coding is also examined: two coding schemes are considered, viz., Ungerboeck's scheme for combination with convolutional codes, and Ginzburg's scheme for combination with block codes. The performance of these schemes makes them attractive for transmission over bandlimited digital channels.

This report summarizes the outcomes of the project outlined ENHANCING THE BANDWIDTH UTILIZATION IN THE MILLIMETER-WAVEBAND AND MODERNIZING THE DIGITAL SIGNAL PROCESSING LABORATORY. Throughout this project we have madesignificant accomplishments on both advancing the state of the art on millimeter-wave communications by developing new transmissionmechanism, network models, and channel characterizations.

This final report covers research carried out under Contract DAAH-04-96-1-0161 for the period June 1, 1996 through September 30, 1999. Research activities are concentrated in the following areas: multidimensional digital signal processing, optical storage and information processing, and electromagnetic measurements. Individual work units involve research in multidimensional digital signal processing and modeling, stereo and multiview image processing, nonlinear systems for image and video signal processing, linear multidimensional multiresolution processing, optical devices for information processing, semiconductor quantum wave devices, electromagnetic analysis and measurements in the time and frequency domains, and far-field, compact and near-field antenna measurement facility compensation.

From the bitsavers.org collectio n, a scanned-in computer-related document. components :: harris :: 1994 Harris Digital Signal Processing Data Book

Creating an embedded system that meets its functional, performance, cost, and schedule goals is a software-and-hardware codesign problem, since the design of the software and hardware components influence each other. The traditional design methodology is sequential, with hardware designed first and then software. The lack of a unified and unbiased approach can lead to suboptimal design and incompatibilities across the software and hardware boundary. To solve these problems, we propose a new software/firmware/hardware codesign methodology to systematically build correct designs efficiently. This codesign methodology includes requirements development, architecture forming, software/ firmware/hardware partitioning, design-pattern mapping, new-design pattern synthesis, integration, and testing. We tested our methods on three application areas. One was a digitizer-filter architecture for ultra-high frequency signals for which we synthesized design patterns in firmware to meet high-frequency requirements. Another was a digitizer-filter architecture for low-frequency signals. A third was a hidden Markov model using dynamic programming. We implemented and tested the first application on a Tektronix/Synopsys embedded system and the second on a Pentek embedded system based on the requirements provided by the stakeholders

An existing constant-temperature hot-wire anemometer system that enabled broadband-turbulence quantities to be determined using a hybrid analog/digital measurement technique has been modified so that turbulence quantities can now be determined using a purely digital technique. With the new system, higher-degree turbulence terms, such as u3, V3, w3, u2v, uv2, u4, v4 and w4, can now be obtained. These terms are used to calculate turbulent-kinetic-energy and Reynolds-shear-stress balances, and skewness and flatness factors, that could not be calculated previously. In this report the new equipment is described and the procedures and computer programs used to take measurements and process the data are detailed.

Video signal processor uses special-purpose integrated circuits with nonsaturating current mode switching to accept texture and color information from a digital computer in a visual spaceflight simulator and to combine these, for display on color CRT with analog information concerning fading.

A radiation tolerant testbed was designed using a Commercial-Off-the- Self Digital Signal Processor and presented to prove the concept of Triple Modular Redundant (TMR) processors in order to make a COTS DSP radiation tolerant design. The system was designed to handle the effects of radiation associated with Single Event Upset only. Two of the industry's leading programmable 32-bit floating-point digital signal processors were reviewed for this thesis, Analog Devices ADSP-21060 and the Texas Instruments TMS320C6701. The 6701 was the best processor for this design based upon size, power, speed, and tolerance to single event latchup, signal event burnout, and total ionization dose. A review of the processor's performance and characteristics is provided to ensure the proper operation of 6701 in a TMR design. The system employs a bit by bit voter that compares the three processors' results and outputs the majority of the bits. All data, address, and control signals are monitored to determine that the system is operating properly. This system significantly differs from previous TMR designs, because only address errors cause immediate interrupts. Data errors cause processor interrupts only when the errors accumulate to a critical level. An external host processor controls the processors' shared memory space.

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The research sponsored by this grant is focused on the development of a theoretical and technological basis for designing efficient systolic arrays for digital-signal processing algorithms. The major contributions of the research are the following: Data reduction techniques via the utilization of algorithm properties Conversion of sequential input signal into input blocks by means of a spiral systolic mesh that is suitable for parallel processing and that is flexible for enabling various array dimensions Devising a hybrid of SA and data-flow approach, making data streams independent of computations executed in each processor, thus reducing waiting time. The communication (PE) protocols resolve the data-flow conflicts created by the merging of the spiral and asynchronous systolic array architecture.

A graphical software user interface for a VMEbus-based real-time digital signal processing system was designed. User requirements were defined and the Rumbaugh object-oriented analysis and design technique was applied to analyze the requirements and produce an object-oriented design. The software design includes a graphical, mouse- and keyboard-driven user interface, specialized hardware driver modules, and operating system interfaces. An implementation plan was also developed to map the design into the C programming language using existing system code, automatically generated code, and newly written code. Based on the implementation plan, a limited software system prototype was successfully developed and demonstrated. The system can be used to analyze signals previously recorded on disk or sampled in real time. Analyzed signals can be displayed either as a set of discrete samples or as a graph in either the time or frequency domains. The system includes real-time sampling hardware, specialized DSP hardware, general-purpose computing hardware based on the Motorola 68030 microprocessor, mass storage media, and a high-resolution graphical display. Software engineering, Interfaces, Signal processing, Real time, Digital computers, Digital communications.

The general design approach of hardware and software systems and the present performance capability of a digital signal processing laboratory is given. Block diagrams of the hardware and software systems are provided along with an example of a recent task. (Author)

This report describes the technical progress towards the development of signed processing algorithms for hybrid lidar- radar designed to improve detection performance.

Introduction To Digital Signal Processing And Filter Design

Introduction To Digital Signal Processing And Filter Design

We consider the compensation of channel nonlinearities in digital radio communication systems. A discrete system with memory, inserted between the source and the modulator, is designed with the aim of providing an equivalent channel with a distortionless linear part and no nonlinearities up to a given order. This design is based on a Volterra series model of the channel, and on the theory of p-th order inverse systems. Since the compensator design is based on a mathematical model of the channel, the problem of model identification is considered. A modeling technique is described, based on computer simulation and application of orthogonal Volterra series. Several examples show the performance of this class of compensators.

Digital signal processing techniques expand the capabilities of modern radars. The central algorithm of radar signal processing is the fast Fourier transform (FFT). However, each radar has several modes and in most cases, environmental factors such as noise and clutter background are not completely understood. For this reason, flexibility of the signal processor is desirable. A method of gaining this flexibility is via general purpose (i.e., programmable) digital signal processing computer structures. In this note, a variety of such structures, both programmable, yet suitable for high speed FFT, are expanded.

Digital signal processing techniques expand the capabilities of modern radars. The central algorithm of radar signal processing is the fast Fourier transform (FFT). However, each radar has several modes and in most cases, environmental factors such as noise and clutter background are not completely understood. For this reason, flexibility of the signal processor is desirable. A method of gaining this flexibility is via general purpose (i.e., programmable) digital signal processing computer structures. In this note, a variety of such structures, both programmable, yet suitable for high speed FFT, are expanded.

The research supported by this ONR grant has investigated modern high-speed signal processing system design. It has encompassed a complete spectrum of activities, starting with the discovery of new signal processing algorithms, and continuing through the development of the most appropriate methods for their realization, including, in particular, the design, layout and fabrication of integrated circuits. The primary project for this grant has been the design and implementation of a new type of programmable general purpose digital filter IC. It employs multiple processing units on a single integrated circuit. The multiple processors operate in parallel and communicate with one another through on-chip dual-access storage register blocks, thereby incurring no operating speed penalties as would result if it were necessary to read and write to off-chip RAM. The system's topology has the processors arranged in a ring, with locally-shared register blocks between each adjacent pair of processors. Our prototype IC has five processors, and it is capable of realizing a rich variety of filter structures that operate at the maximum instruction execution rate possible for any custom parallel implementation.

The research supported by this ONR grant has investigated modern high-speed signal processing system design. It has encompassed a complete spectrum of activities, starting with the discovery of new signal processing algorithms, and continuing through the development of the most appropriate methods for their realization, including, in particular, the design, layout and fabrication of integrated circuits. The primary project for this grant has been the design and implementation of a new type of programmable general purpose digital filter IC. It employs multiple processing units on a single integrated circuit. The multiple processors operate in parallel and communicate with one another through on-chip dual-access storage register blocks, thereby incurring no operating speed penalties as would result if it were necessary to read and write to off-chip RAM. The system's topology has the processors arranged in a ring, with locally-shared register blocks between each adjacent pair of processors. Our prototype IC has five processors, and it is capable of realizing a rich variety of filter structures that operate at the maximum instruction execution rate possible for any custom parallel implementation.

We have continued our development of the 12-bit by 11-bit multiplier for our multiprocessor-ring chip. Since the multiplier's performance is the single most important factor in determining the maximum speed at which our system can operate, it is essential that this task be performed in the best possible manner. Our goal was that this multiplier operate in approximately 30 ns, so that our overall chip speed could reach 30 MHz. The multiplier was designed to accommodate eleven-bit data and twelve-bit coefficients. In addition to the above-cited operating speed goal, we needed to make the multiplier small enough that five complete processors, including five multipliers, can be included on a single IC (which employs 2-microns CMOS technology). As mentioned in the previous progress report, after completing the multiplier's layout and verifying its correct functioning via logic simulation, we performed SPICE simulations on the various multiplier critical-path components. These simulations indicated an operating speed well under 30 ns. We next fabricated a MOSIS TinyChip (2.22 mm x 2.25 mm, in 2-micron technology) with this 12-bit by 11-bit multiplier and a small block of coefficient RAM as well as a small dual- port register block. this allowed us to test the multiplier's performance in an environment that closely simulated the five-processor-ring system.

A radiation tolerant testbed was designed using a Commercial-Off-the-Self Digital Signal Processor and presented to prove the concept of Triple Modular Redundant (TMR) processors in order to make a COTS DSP radiation tolerant design. The system was designed to handle the effects of radiation associated with Single Event Upset only. Two of the industry's leading programmable 32-bit floating-point digital signal processors were reviewed for this thesis, Analog Devices ADSP-21060 and the Texas Instruments TMS320C6701. The 6701 was the best processor for this design based upon size, power, speed, and tolerance to single event latchup, signal event burnout, and total ionization dose. A review of the processor's performance and characteristics is provided to ensure the proper operation of 6701 in a TMR design. The system employs a bit by bit voter that compares the three processors' results and outputs the majority of the bits. All data, address, and control signals are monitored to determine that the system is operating properly. This system significantly differs from previous TMR designs, because only address errors cause immediate interrupts. Data errors cause processor interrupts only when the errors accumulate to a critical level. An external host processor controls the processors' shared memory space.

This paper presents a prediction error variance reduction procedure based on fractional digital differentiation with negative order. This reduction is achieved by increasing correlation in the signals. Applications to ECG signals show that savings of more than one bit per residual signal sample can be attained.

Backscatter reduction is a critical stage in enhancing the usable range of any hybrid lidar-radar scheme that is deployed into a turbid underwater environment. In a highly scattering environment, many photons reaching the detector will have scattered off particulates in the water, while relatively few photons reaching the detector wifl have made the round-trip to and from the object of interest. This will cause the system to detect an object whose range is near the volumetric center of the scattering region, rather than the detecting the range to the object of interest. A similar problem is encountered in the radar areas of moving target indication (MTI) and through-the-wall radar imaging (TWRI), in which radar clutter obscures the desired object return [l,2j. Researchers in these areas have shown that the clutter has lower spatial frequencies than the object, meaning that the clutter return can theoretically be filtered out by a spatial filter without negatively impacting the target return. The backscatter in the turbid underwater environment is analogous to the clutter in these radar scenarios. An early solution to this problem was the use of delay line cancelers, with more recent work focusing on more sophisticated signal processing solutions such as FFT-based filters. The main idea behind the delay line canceler is that the clutter signal is a low frequency signal, which can be attenuated by the high-pass quality of a differentiator. Further modifications can be made to delay line cancelers to fine-tune the behavior, such as widening the clutter rejection region.

Backscatter reduction is a critical stage in enhancing the usable range of any hybrid lidar-radar scheme that is deployed into a turbid underwater environment. In a highly scattering environment, many photons reaching the detector will have scattered off particulates in the water, while relatively few photons reaching the detector wifl have made the round-trip to and from the object of interest. This will cause the system to detect an object whose range is near the volumetric center of the scattering region, rather than the detecting the range to the object of interest. A similar problem is encountered in the radar areas of moving target indication (MTI) and through-the-wall radar imaging (TWRI), in which radar clutter obscures the desired object return [l,2j. Researchers in these areas have shown that the clutter has lower spatial frequencies than the object, meaning that the clutter return can theoretically be filtered out by a spatial filter without negatively impacting the target return. The backscatter in the turbid underwater environment is analogous to the clutter in these radar scenarios. An early solution to this problem was the use of delay line cancelers, with more recent work focusing on more sophisticated signal processing solutions such as FFT-based filters. The main idea behind the delay line canceler is that the clutter signal is a low frequency signal, which can be attenuated by the high-pass quality of a differentiator. Further modifications can be made to delay line cancelers to fine-tune the behavior, such as widening the clutter rejection region.

Instructor: Alan V. Oppenheim Set of 20 video lectures for Signals and Systems, an introductory course in analog and digital signal processing, including seismic data processing, communications, speech processing, image processing, consumer electronics, and defense electronics. View the complete course: http://ocw.mit.edu/RES6-008S11 License: Creative Commons BY-NC-SA More information at http://ocw.mit.edu/terms More courses at http://ocw.mit.edu

Instructor: Alan V. Oppenheim Set of 20 video lectures for Signals and Systems, an introductory course in analog and digital signal processing, including seismic data processing, communications, speech processing, image processing, consumer electronics, and defense electronics. View the complete course: http://ocw.mit.edu/RES6-008S11 License: Creative Commons BY-NC-SA More information at http://ocw.mit.edu/terms More courses at http://ocw.mit.edu

Instructor: Alan V. Oppenheim Set of 20 video lectures for Signals and Systems, an introductory course in analog and digital signal processing, including seismic data processing, communications, speech processing, image processing, consumer electronics, and defense electronics. View the complete course: http://ocw.mit.edu/RES6-008S11 License: Creative Commons BY-NC-SA More information at http://ocw.mit.edu/terms More courses at http://ocw.mit.edu

There is a growing trend toward using high-level tools for design and implementation of radio astronomy digital signal processing (DSP) systems. Such tools, for example, those from the Collaboration for Astronomy Signal Processing and Electronics Research (CASPER), are usually platform-specific, and lack high-level, platform-independent, portable, scalable application specifications. This limits the designer's ability to experiment with designs at a high-level of abstraction and early in the development cycle. We address some of these issues using a model-based design approach employing dataflow models. We demonstrate this approach by applying it to the design of a tunable digital downconverter (TDD) used for narrow-bandwidth spectroscopy. Our design is targeted toward an FPGA platform, called the Interconnect Break-out Board (IBOB), that is available from the CASPER. We use the term TDD to refer to a digital downconverter for which the decmation factor and center frequency can be reconfigured without the need for regenerating the hardware code. Such a design is currently not available in the CASPER DSP library. The work presented in this paper focuses on two aspects. Firstly, we introduce and demonstrate a dataflow-based design approach using the dataflow interchange format (DIF) tool for high-level application specification, and we integrate this approach with the CASPER tool flow. Secondly, we explore the trade-off between the flexibility of TDD designs and the low hardware cost of fixed-configuration digital downconverter (FDD) designs that use the available CASPER DSP library. We further explore this trade-off in the context of a two-stage downconversion scheme employing a combination of TDD or FDD designs.

There is a growing trend toward using high-level tools for design and implementation of radio astronomy digital signal processing (DSP) systems. Such tools, for example, those from the Collaboration for Astronomy Signal Processing and Electronics Research (CASPER), are usually platform-specific, and lack high-level, platform-independent, portable, scalable application specifications. This limits the designer's ability to experiment with designs at a high-level of abstraction and early in the development cycle. We address some of these issues using a model-based design approach employing dataflow models. We demonstrate this approach by applying it to the design of a tunable digital downconverter (TDD) used for narrow-bandwidth spectroscopy. Our design is targeted toward an FPGA platform, called the Interconnect Break-out Board (IBOB), that is available from the CASPER. We use the term TDD to refer to a digital downconverter for which the decmation factor and center frequency can be reconfigured without the need for regenerating the hardware code. Such a design is currently not available in the CASPER DSP library. The work presented in this paper focuses on two aspects. Firstly, we introduce and demonstrate a dataflow-based design approach using the dataflow interchange format (DIF) tool for high-level application specification, and we integrate this approach with the CASPER tool flow. Secondly, we explore the trade-off between the flexibility of TDD designs and the low hardware cost of fixed-configuration digital downconverter (FDD) designs that use the available CASPER DSP library. We further explore this trade-off in the context of a two-stage downconversion scheme employing a combination of TDD or FDD designs.

There is a growing trend toward using high-level tools for design and implementation of radio astronomy digital signal processing (DSP) systems. Such tools, for example, those from the Collaboration for Astronomy Signal Processing and Electronics Research (CASPER), are usually platform-specific, and lack high-level, platform-independent, portable, scalable application specifications. This limits the designer's ability to experiment with designs at a high-level of abstraction and early in the development cycle. We address some of these issues using a model-based design approach employing dataflow models. We demonstrate this approach by applying it to the design of a tunable digital downconverter (TDD) used for narrow-bandwidth spectroscopy. Our design is targeted toward an FPGA platform, called the Interconnect Break-out Board (IBOB), that is available from the CASPER. We use the term TDD to refer to a digital downconverter for which the decmation factor and center frequency can be reconfigured without the need for regenerating the hardware code. Such a design is currently not available in the CASPER DSP library. The work presented in this paper focuses on two aspects. Firstly, we introduce and demonstrate a dataflow-based design approach using the dataflow interchange format (DIF) tool for high-level application specification, and we integrate this approach with the CASPER tool flow. Secondly, we explore the trade-off between the flexibility of TDD designs and the low hardware cost of fixed-configuration digital downconverter (FDD) designs that use the available CASPER DSP library. We further explore this trade-off in the context of a two-stage downconversion scheme employing a combination of TDD or FDD designs.

Learning how to use my new Xiegu X5105. In this video, I'm showing you how I'm using the DSP (Digital Signal Processing) features of this digital transceiver to eliminate the horrendous amount of static and interference on the radio signal. Being a digital radio, it has some excellent noise filtering capabilities. I'm playing with Attenuation, Pre-Amp, Noise Reduction, Audio Filter (Low pass/High Pass), Notch Filter. I haven't used the Noise Blanker feature since it's for ignition type noise. No effect. If you're studying for your HAM license test, you will actually see these terms used in practice so it will be easier to remember. I'm an Amateur Radio operator - General Class. Just getting started on HF so learn it with me. I'm the Internet Privacy Guy. I'm a public interest technologist. I'm here to educate. You are losing your Internet privacy and Internet security every day if you don't fight for it. Your data is collected with endless permanent data mining. Learn about a TOR router, a VPN , antivirus, spyware, firewalls, IP address, wifi triangulation, data privacy regulation, backups and tech tools, and evading mass surveillance from NSA, CIA, FBI. Learn how to be anonymous on the Internet so you are not profiled. Learn to speak freely with pseudo anonymity. Learn more about the dangers of the inernet and the dangers of social media, dangers of email. I like alternative communication technology like Amateur Radio and data communications using Analog. I'm a licensed HAM operator. Contact Rob on the Brax.Me App (@robbraxman) for encrypted conversations. https://bytzVPN.com Premium VPN with Cloud-Based TOR Routing https://whatthezuck Cybersecurity Reference https://brax.me Privacy Focused Social Media - Encrypted Communications http://rob.brax.me Store for BytzVPN, BraxWifi Router, and merchandise

The engineering development study to follow was written to address the need for a Programmable Rate Digital Satellite Modem capable of supporting both burst and continuous transmission modes with either binary phase shift keying (BPSK) or quadrature phase shift keying (QPSK) modulation. The preferred implementation technique is an all digital one which utilizes as much digital signal processing (DSP) as possible. Here design tradeoffs in each portion of the modulator and demodulator subsystem are outlined, and viable circuit approaches which are easily repeatable, have low implementation losses and have low production costs are identified. The research involved for this study was divided into nine technical papers, each addressing a significant region of concern in a variable rate modem design. Trivial portions and basic support logic designs surrounding the nine major modem blocks were omitted. In brief, the nine topic areas were: (1) Transmit Data Filtering; (2) Transmit Clock Generation; (3) Carrier Synthesizer; (4) Receive AGC; (5) Receive Data Filtering; (6) RF Oscillator Phase Noise; (7) Receive Carrier Selectivity; (8) Carrier Recovery; and (9) Timing Recovery.

Topics covered involve a number of closely related tasks including: the development of several control loop and dynamic noise model computer programs for simulating microwave radiometer measurements; computer modeling of an existing stepped frequency radiometer in an effort to determine its optimum operational characteristics; investigation of the classical second order analog control loop to determine its ability to reduce the estimation error in a microwave radiometer; investigation of several digital signal processing unit designs; initiation of efforts to develop required hardware and software for implementation of the digital signal processing unit; and investigation of the general characteristics and peculiarities of digital processing noiselike microwave radiometer signals.

Topics covered involve a number of closely related tasks including: the development of several control loop and dynamic noise model computer programs for simulating microwave radiometer measurements; computer modeling of an existing stepped frequency radiometer in an effort to determine its optimum operational characteristics; investigation of the classical second order analog control loop to determine its ability to reduce the estimation error in a microwave radiometer; investigation of several digital signal processing unit designs; initiation of efforts to develop required hardware and software for implementation of the digital signal processing unit; and investigation of the general characteristics and peculiarities of digital processing noiselike microwave radiometer signals.

We address the problem of discretizing continuous cosmological signals such as a galaxy distribution for further processing with Fast Fourier techniques. Discretizing, in particular representing continuous signals by discrete sets of sample points, introduces an enormous loss of information, which has to be understood in detail if one wants to make inference from the discretely sampled signal towards actual natural physical quantities. We therefore review the mathematics of discretizing signals and the application of Fast Fourier Transforms to demonstrate how the interpretation of the processed data can be affected by these procedures. It is also a well known fact that any practical sampling method introduces sampling artifacts and false information in the form of aliasing. These sampling artifacts, especially aliasing, make further processing of the sampled signal difficult. For this reason we introduce a fast and efficient supersampling method, frequently applied in 3D computer graphics, to cosmological applications such as matter power spectrum estimation. This method consists of two filtering steps which allow for a much better approximation of the ideal sampling procedure, while at the same time being computationally very efficient.Thus, it provides discretely sampled signals which are greately cleaned from aliasing contributions.

We address the problem of discretizing continuous cosmological signals such as a galaxy distribution for further processing with Fast Fourier techniques. Discretizing, in particular representing continuous signals by discrete sets of sample points, introduces an enormous loss of information, which has to be understood in detail if one wants to make inference from the discretely sampled signal towards actual natural physical quantities. We therefore review the mathematics of discretizing signals and the application of Fast Fourier Transforms to demonstrate how the interpretation of the processed data can be affected by these procedures. It is also a well known fact that any practical sampling method introduces sampling artifacts and false information in the form of aliasing. These sampling artifacts, especially aliasing, make further processing of the sampled signal difficult. For this reason we introduce a fast and efficient supersampling method, frequently applied in 3D computer graphics, to cosmological applications such as matter power spectrum estimation. This method consists of two filtering steps which allow for a much better approximation of the ideal sampling procedure, while at the same time being computationally very efficient.Thus, it provides discretely sampled signals which are greately cleaned from aliasing contributions.

Digital Signal Processing Applications - Using the ADSP-2100 Family IBM PC / 3.5" Contains scans  and dump  of the software. Scans are 1200  dpi RAW w/ ICC profile embedded Lossless JPEG XL w/ diffusion layer where useful   Includes Disk - Label Book - Cover These scans are for preservation, not presentation. and need additional processing like inverse halftoning, cropping and straightening to look presentable. Dump is Kryoflux Raw

We present a digital signal processing system based on a photon counting approach which we developed for a thermal neutron detector consisting of ZnS(Ag):6LiF scintillating layers read out with WLS fibers and SiPMs. Three digital filters have been evaluated: a moving sum, a moving sum after differentiation and a digital CR-RC^4 filter. The performances of the detector with these filters are presented. A full analog signal processing using a CR-RC^4 filter has been emulated digitally. The detector performance obtained with this analog approach is compared with the one obtained with the best performing digital approach.

From the bitsavers.org collectio n, a scanned-in computer-related document. components :: harris :: 1991 Harris Digital Signal Processing

From the bitsavers.org collectio n, a scanned-in computer-related document. harris :: dataBooks :: 1993 Harris Digital Signal Processing Databook

This paper explores finding the number $n_h$ of undirected hamiltonian paths in an undirected graph $G=(V,E)$ using lumped/ideal circuits, specifically low-pass filters. Ideal analog computation allows one to computer $n_h$ in a short period of time, but in practice, precision problems disturb this ideal nature. A digital/algorithmic approach is proposed, and then it is shown that the approach/method operates under theoretically feasible (polynomial) time: $O(n^{230})+O(n_d PC(p_2,n_d))$, where $n=|V|$ and $PC(p_2,n_d)$ refers to the time complexity of finding a root of a polynomial with polynomial coefficients being $p_2$ digits (with coefficients limited within $2^{p_2/2}$ and $1/2^{p_2/2}$ in magnitude) and $n_d$ referring to number of degree, with $p_2 =n^{50}$ and $n_d = n^{10}$.

This paper explores finding the number $n_h$ of undirected hamiltonian paths in an undirected graph $G=(V,E)$ using lumped/ideal circuits, specifically low-pass filters. Ideal analog computation allows one to computer $n_h$ in a short period of time, but in practice, precision problems disturb this ideal nature. A digital/algorithmic approach is proposed, and then it is shown that the approach/method operates under theoretically feasible (polynomial) time: $O(n^{230})+O(n_d PC(p_2,n_d))$, where $n=|V|$ and $PC(p_2,n_d)$ refers to the time complexity of finding a root of a polynomial with polynomial coefficients being $p_2$ digits (with coefficients limited within $2^{p_2/2}$ and $1/2^{p_2/2}$ in magnitude) and $n_d$ referring to number of degree, with $p_2 =n^{50}$ and $n_d = n^{10}$.

This paper explores finding the number $n_h$ of undirected hamiltonian paths in an undirected graph $G=(V,E)$ using lumped/ideal circuits, specifically low-pass filters. Ideal analog computation allows one to computer $n_h$ in a short period of time, but in practice, precision problems disturb this ideal nature. A digital/algorithmic approach is proposed, and then it is shown that the approach/method operates under theoretically feasible (polynomial) time: $O(n^{230})+O(n_d PC(p_2,n_d))$, where $n=|V|$ and $PC(p_2,n_d)$ refers to the time complexity of finding a root of a polynomial with polynomial coefficients being $p_2$ digits (with coefficients limited within $2^{p_2/2}$ and $1/2^{p_2/2}$ in magnitude) and $n_d$ referring to number of degree, with $p_2 =n^{50}$ and $n_d = n^{10}$.