Realization of computers using programmable logic units

The programmable logic unit (PLU) has been proposed to realize a high‐speed operation. The idea is to perform computation obtained by a static data flow scheme using the stepwise arithmetic logics distributed in the memory. This paper describes the configuration of the prototype PLU, the basic concepts of the computation and the synchronization between the PLU and the main control unit (CU). Then theoretical discussions are made on the computation time for the arithmetic and the iterative programs. It is shown that the high‐speed processing is realized by the effects of the parallel computation and of the pipelined computation, mapping many templates of a loop program on the PLU. Furthermore, the multiprogramming for the computer with PLU is proposed to utilize effectively the waiting time for the computation in the PLU. The throughput improvement is indicated by theoertical analysis, comparing the computer using PLU with the traditional sequential computer. Finally, the usefulness of the PLU is demonstrated.

Robotic manufacturing cells are heterogeneous systems consisting of such different components as machine tools, robots and programmable logic units. The efficient management of a cell depends on the integration of these different elements. This paper discusses the challenging problems and perspectives relating to the use of advanced communication and programming methodologies which rely on the local area network as the interconnection structure for the cell components.

Data processing is one of the essential techniques to obtain accurate responses of the earth. The Fast Fourier Transform (FFT) has been widely used in the processing of magnetotelluric and other electromagnetic data, and gives us electromagnetic response functions when the signal-to-noise ratio is high or if the length of time series is long enough. However, these conditions may fail from time to time depending on the environmental conditions of exploration. We now focus on the data processing techniques using not the FFT but digital filters that have been “proven” effective to process seismic data. Such data processing techniques have been rarely applied to electromagnetic data. In this study, we apply a digital filter called “pole on pedestal” that amplifies or extracts signals at a specific frequency with less distortion of the Fourier spectrum than those obtained by the FFT. We first elucidate how effectively the application of digital filters improves the quality of electrical imaging using “pole on pedestal.” Next, we then apply this filter to marine electrical and electromagnetic survey data to demonstrate the effectiveness of the method. Real marine data-sets acquired by a deep-towed DC resistivity survey system and ocean-bottom instruments are used. It is obviously shown that the pole-on-pedestal filter improves the quality of the response functions and the estimated sub-seafloor resistivity structure.

Fiber alignment plays a critical role in the structure and function of cells and tissues. While fiber alignment quantification is important to experimental analysis and several different methods for quantifying fiber alignment exist, many studies focus on qualitative rather than quantitative analysis perhaps due to the complexity of current fiber alignment methods. Speed and sensitivity were compared in edge detection and fast Fourier transform (FFT) for measuring actin fiber alignment in cells exposed to shear stress. While edge detection using matrix multiplication was consistently more sensitive than FFT, image processing time was significantly longer. However, when MATLAB functions were used to implement edge detection, MATLAB's efficient element-by-element calculations and fast filtering techniques reduced computation cost 100 times compared to the matrix multiplication edge detection method. The new computation time was comparable to the FFT method, and MATLAB edge detection produced well-distributed fiber angle distributions that statistically distinguished aligned and unaligned fibers in half as many sample images. When the FFT sensitivity was improved by dividing images into smaller subsections, processing time grew larger than the time required for MATLAB edge detection. Implementation of edge detection in MATLAB is simpler, faster, and more sensitive than FFT for fiber alignment quantification.

This paper presents design algorithms for hybrid filter banks (HFBs) for high-speed, high-resolution conversion between analog and digital signals. The HFB is an unconventional class of filter bank that employs both analog and digital filters. When used in conjunction with an array of slower speed converters, the HFB improves the speed and resolution of the conversion compared with the standard time-interleaved array conversion technique. The analog and digital filters in the HFB must be designed so that they adequately isolate the channels and do not introduce reconstruction errors that limit the resolution of the system. To design continuous-time analog filters for HFBs, a discrete-time-to-continuous-time ("Z-to-S") transform is developed to convert a perfect reconstruction (PR) discrete-time filter bank into a near-PR HFB; a computationally efficient algorithm based on the fast Fourier transform (FFT) is developed to design the digital filters for HFBs. A two-channel HFB is designed with sixth-order continuous-time analog filters and length 64 FIR digital filters that yield -86 dB average aliasing error. To design discrete-time analog filters (e.g., switched-capacitors or charge-coupled devices) for HFBs, a lossless factorization of a PR discrete-time filter bank is used so that the reconstruction error is not affected by filter coefficient quantization. A gain normalization technique is developed to maximize the dynamic range in the finite-precision implementation. A four-channel HFB is designed with 9-bit (integer) filter coefficients. With internal precision limited to the equivalent of 15 bits, the maximum aliasing error is -70 dB, and with the equivalent of 20 bits internal precision, maximum aliasing is -100 dB. The 9-bit filter coefficients degrade the stopband attenuation (compared with unquantized coefficients) by less than 3 dB.

To reduce the processing time in Global Navigation Satellite System (GNSS) signal acquisition and tracking, we first propose a parallel method of signal acquisition and tracking as well as the associated software implementation in this paper. This paper applies the Message Passing Interface (MPI) and Open Multi-Processing (OpenMP) techniques for parallel computation implementation. This parallel method could put the multi-cores microprocessors to be efficiently used in GNSS signal acquisition and tracking processes. To further enhance the performance, this paper then proposes a new GNSS acquisition and tracking method, and it includes the multi-PRN code acquisition method we proposed in [Y. C. Lin and C. C. Sun, ION GNSS 2007] to reduce the acquisition processing time and a new FFT (Fast Fourier Transform)-based tracking method. This FFT-based tracking method combines the conventional tracking method and the FFT technique to track the incoming signal. Consequently, the tracking process is divided into two phases: detection and FFT-based phases. This proposed FFT-based tracking method can reduce the computational load as well. This new acquisition and tracking method will be combined with the above parallel method to gain the maximum improvement in GNSS signal acquisition and tracking. Finally, we use the real data collected by a software GNSS receiver to validate the proposed method. This paper will also conduct the comparison between the proposed method and the conventional GNSS acquisition and tracking method, and the comparison criterions are the computational load and the processing time. The experiment results of this paper show the effectiveness of this proposed approach for GNSS signal acquisition and tracking.

Digital filters are indispensible element in digital signal processing area. The performance of digital filter based on adding and multiplying operation, such as computational speed and power consuming is determined by the orders and coefficients of filter which has on effect area of semiconductor chip when it is implemented by VLSI technology. In this research, in order to performance improvement of digital filter, we proposed the algorithm to speed-up the operation of digital filter associated with the minimum signed digit representation of binary number system and method to simplify the digital filter design associated with common subexpression elimination. The performance of proposed method is evaluated by the computational speed and design-simplicity by experimental implemented digital filter on FPGA.

The spectrum of a musical signal contains features that are important in our perception of musical sounds. We know that the timbre or tone quality depends, in part, on the spectrum of the tone (Grey:1978). Also, the location of prominent frequency components in the spectrum have been shown to relate to the perceived pitch of the tone as well (Piszczalski and Galler: 1978). Unfortunately, a single spectrum provides no information on the time-varying aspects of sound. Consequently, methods have been devised to capture and display the time-varying spectrum of sounds. Perhaps the best known display of amplitude -frequencytime information is in spectrograms, more commonly known as voiceprints. The speech community has long used the analog spectrograph to generate these images, and musical examples were studied using this technique as early as 1947 (Potter et al.). More recently digital methods have been employed to capture the spectrographic image. Both digital filters and the Fast Fourier Transform (FFT) can give spectrographic information, with the FFT being the method used in the figures presented here. For more information on the actual computer implementation cf. (Piszczalski and Galler: 1978.) For spectrographic analysis, the digital computer has advantages over the analog spectrograph by providing more sophisticated graphics displays, including a three-dimensional representation, which we refer to as the surface of the sound. Also, digitized data is more amenable to further analysis and processing. Spectrographic displays in general offer a unique graphic perspective for music acoustics research; in particular, in studying sounds produced on traditional musical instruments. We can study how the harmonic envelope changes during the course of a single sustained tone. We can also view how the spectrum changes between played notes. We can also watch the effect of articulation, such as legato-tonguing, on the resulting spectrum. Spectrograms also reduce the danger of incorrectly assuming that an arbitrary spectrum is representative for all tones produced on that instrument. The variety of shapes the harmonic envelope may take can be especially dramatic when a sequence of notes is simultaneously displayed, as is often the case in the following figures. Depending on what perceptual features are sought, different frequency and time scales should be used. We wanted the melodic patterns to be as visually obvious as possible, so we optimized towards identifying the note sequences. In the following spectral surfaces a new spectrum was calculated for every 32 msec of music. We have found this time interval to be sufficiently dense to capture the most rapidly played note sequences that we have studied to date (up to 14 notes/sec). The frequency scale is subdivided into 128 equally spaced points between 0 Hz and the maximum frequency indicated on the respective graphs. The hidden-line, three-dimensional algorithm used for our graphics was implemented on a Hewlett-Packard minicomputer system by Frederick Looft. The frequency, amplitude and time scales are linear in all cases. These images were generated in the course of our larger study of musical psychoacoustics, and more specifically have served as an aid in the areas of automatic pitch-tracking and melodic pattern recognition of performed music. While we have found the spectral representations quite helpful, many of these graphs have generated fascinating questions which we simply have not had time yet to explore to any depth or to evaluate using other techniques. Hearing the source of the sounds while viewing the figures makes them particularly effective. Still, this collection of sound images, by themselves, should be informative for those in the music analysis/synthesis area. Except where otherwise noted, audio sources were recorded at the University of Michigan. This research is supported by the National Science Foundation (Grant No. MC578-09052).

Refractive index profile of surface channel waveguides can be determined by analyzing the near-field intensity pattern. A mathematical model, the inverse Helmholtz equation, is derived in order to use these data to reconstruct the refractive index profile under consideration. In this work, the measured near-field intensities are preprocessed by means of gamma correction, background noise subtraction, Fast Fourier Transform, and low pass finite impulse response (FIR) digital filter. Several types of FIR windows are chosen. The results are used to reconstruct the refractive index profile of the waveguide. The results show that the application of low pass FIR digital filter by using Hamming window reduces noises better than other windows. The application of this method in determination of refractive index profile of annealed proton exchange LiNbO<SUB>3</SUB> channel waveguides is demonstrated.

The utilization of open and free-environment programming has increased drastically in education and research fields in the current pandemic situation. SCILAB programming originated from a laboratory INRIA in France and it is the overall open and free-environment reference in mathematical calculation programming. It is an alternative open-source with an easy syntax to the licensed software MATLAB. This preliminary study paper describes about the design and simulation of the discrete-time signal processing laboratory concepts with the SCILAB open-source software. The transform techniques in signal processing like Fourier Transform of discrete signals, fast Fourier Transform, Fourier series for discrete-time periodic signals and digital filters are better understood with the simulation results obtained in SCILAB. The student feedback shows that the student attention has improved with SCILAB based signal processing teaching.

In this paper we proposed a new algorithm that can reduce the open-loop pitch estimation time of G.723.1. The time domain cross-correlation method is simple but has long processing time by recursive multiplication. For reduction of processing time, we use the method that compute the cross-correlation by multiplying the Fourier value of speech by it's complex conjugate. Also, we can reduce the processing time by omitting the bit-reversing of FFT and IFFT for time-frequency domain transform. As a result, the processing time of improved hybrid domain cross-correlation algorithm is reduced by 67.37% of conventional time domain cross-correlation.

PURPOSE: Electromagnetic (EM) kinematic tracking systems have gained popularity in recent years due to their ability to quickly capture 6 degree-of-freedom position data. EM systems work by sequentially creating orthogonally oriented magnetic fields in which the receivers locate themselves. The purpose of this investigation was to determine the impact of the EM signal on surface EMG data and to provide suggested methods for reducing any detrimental effects. EQUIPMENT / METHODS: EM signals are produced by EM kinematic tracking systems. The EM field-strength varies as a function of the distance from the transmitter. The frequency varies as a function of the selected kinematic sampling rate. These combine to produce a non-stationary signal that can contaminate EMG data. Raw EMG data was collected from the quadriceps and hamstring muscles on subjects performing a squatting task and a jumping task in the presence of an EM transmitter. Both tasks were performed with and without the transmitter on. Data were collected at 1000 Hz. ANALYSIS Data from all trials were qualitatively assessed for the appearance of EM contamination. These data were then processed to evaluate the impact of the EM signal on muscle timing analyses. Power spectrums of the data were computed using a Fast Fourier Transform (FFT) algorithm. This spectral information was used to create digital filters to remove the EM signal from the raw data. RESULTS The EM signal was present in the EMG data for all trials performed in the presence of an active EM transmitter. In many cases, this signal was strong enough to completely mask the EMG signal. The muscle timing analysis of contaminated data were unable to accurately quantify onsets. Digital filtering successfully removed the EM noise and provided muscle latencies that were comparable to those found from the analysis of the uncontaminated data. CONCLUSIONS Digital filtering is an effective way of removing EM contamination from EMG data. When using this method, care should be taken to collect raw EMG data at a sampling rate that is consistent with the digital filtering requirements. RELEVANCE Extraneous EM signals have historically been part of EMG data collection. Many commercial systems have built-in notch filters to contend with 60Hz interference. The advent of EM tracking systems produces a need to re-evaluate processing techniques for EMG data collected in these environments.

Chapter 6 - Fast Fourier transform methods

The computational complexity of FFT (Fast Fourier Transform) processing in an OFDM (Orthogonal Frequency Division Multiplex) based receiver is large with a large amount of receiver antennas. In LTE (Long Term Evolution) release 8, FFTs of different size are used for user data and random-access preambles [1] requiring additional FFTs to be implemented for random-access reception. Within the current paper, a 5G random-access preamble format is proposed based on a short sequence of the same length as the length of the OFDM symbols that are used for other uplink physical channels, such as user data, control signaling, and reference signals. The preamble sequence is constructed by repeating the short sequence multiple times. A corresponding preamble detector in which FFTs of the same size as for other uplink channels and signals are used is also described. In this way, the amount of special random-access related processing and hardware support is significantly reduced for multi-antenna systems with frequency-domain beamforming. This preamble detector is also robust against inter-carrier interference from other uplink channels and signals. Furthermore, the proposed preamble detector scheme can be used in 5G scenarios with a high amount of phase noise and frequency errors. For time-domain beamforming, the beamforming weights can be changed during preamble reception such that the number of spatial directions is increased for which preamble detection is done. Simulation results are used to compare preamble formats for different lengths of the sequences.

The development of VLSI and LSI technology for telecommunications has been aided by the large volume of integrated circuits that find application in telecommunication market. Using the PCM filter design as a case study, in this paper, we review the developments of LSI and VLSI technology in the past two decades and the manner in which these developments have been utilized. Extensive coverage on analog designs, switched-capacitor designs, and more recent digital filter designs is presented.