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Four operands Real Multiply Accumulation (4op-RMA) is a method to implement transversal circuits, direct form circuits and 2D type circuits with real coefficients by using a complex multiplier. Digital Signal Processors (DSPs) with 4op-RMA implement these circuits twice as fast as the conventional ones. Besides, we can simplify the programs on such DSPs. However 4op-RMA have not been applied to the lattice circuits yet because of their circuit structures. In this paper, we propose an implementation method for the lattice circuits by using 4op-RMA. The lattice part can be calculated by a 4op-RMA operation, which corresponds to a complex multiplication.

This paper gives a simple proof for the positiveness of two important symmetric Toeplitz matrices used in communication and signal processing. It utilizes the shifting property of a so-called Uniformly Band-Restricted (UBR) function, which is the generating function for a generic functional symmetric matrix. It is shown that the functional symmetric matrix is positive definite if the UBR function is evaluated at a sequence of distinct real numbers.

It is important in digital signal processing (DSP) architectures to minimize the silicon area of the integrated circuits. This can be achieved by reducing the number of functional units such as adders and multipliers. In literature, folding technique is used to reduce the functional units by executing multiple algorithm operations on a single functional unit. Register minimization techniques are used to reduce the number of registers in a folded architecture. Retiming is a technique that needs to be performed before applying folding. In this paper, retiming is performed based on nature inspired evolutionary computation method. This technique generates the database of solutions from which best solution can be picked for folding further.

As a part of this work, an efficient folded noise removal audio filter prototype is designed as an application example using evolutionary computation-based retiming and folding with register minimization. Folding technique will however increase the number of registers while multiplexing datapath adder and multiplier elements. Register minimization technique is used after folding to reduce the number of registers. After obtaining retimed, folded filter architecture, low level synthesis is performed which involves mapping of datapath adder and multiplier blocks to actual hardware. Various architectures of adders and multipliers are compared in area-power-performance space and depending on the user defined constraint, folded architecture with specific combination of data path elements is mapped on to hardware. A framework is designed in this paper to automate the entire process which reduces the design cycle time. All the designed filters are targeted for ASIC implementation. The results are compared and are provided as part of simulation results.

This research paper presents a new evolutionary technique named vortex search optimization (VSO) to design digital 2D finite impulse response (FIR) filter for improved performance both in pass-band and stop-band regions. Optimum filter coefficients are calculated by minimizing the deviation of actual frequency response from specified or desired response. Efficiency of the designed filter is measured by several parameters, such as maximum pass-band ripple, maximum stop-band ripple, mean attenuation in stop band and time taken, to execute the code. Analysis of the performance of designed filter is correlated with various different algorithms like real coded genetic algorithm, particle swarm optimization, genetic search algorithm and hybrid particle swarm optimization gravitational algorithm. Comparative study shows significant reduction in pass-band error, stop-band error and execution time.

This Letter shows some counter-intuitive simulation results that for some filter parameters in the extended boundaries of the stability triangle, the state vector will converge to a periodic orbit after some iterations, no matter what the initial conditions. Also, a new pattern, which looks like a rotated letter "X", is found. The center of the rotated letter is located at the origin and the slopes of the "straight lines" of the rotated letter are equal to the values of the pole locations.

This letter shows some counter-intuitive simulation results that the symbolic sequences and the state variables of a digital filter with two's complement arithmetic and arbitrary initial conditions and order will be eventually zero when all the filter parameters are even numbers, no matter the system matrix of the filter is stable or not.

1/f noise is present in many natural and artificial systems, and even though it was discovered decades ago, it is still not completely understood. Due to the lack of an universal model, the main methods of investigating a system where 1/f noise is present are numerical simulations and real measurements. The second method can lead to more adequate results, since it is free from numerical artifacts. In the case of real measurements, we need reliable, wide-band noise generators. Many ways of generating noise are known; most of them have several limitations on the frequency bandwidth or on spectral properties. We wanted to create a device which is easy to use, which can generate any kind of 1/f^a (0 < a < 2) noise and whose bandwidth is wide enough to make our investigations. We used a DSP (ADSP2181) to numerically generate the desired noise, and a D/A converter to convert it to an analogue signal. The noise generation algorithm was based on the known method of filtering a Gaussian white noise with a series of first-order digital filters. We enhanced this method to get a better spectral shape and to compensate for the side effects of the digital-to-analogue conversion.

Acoustic feedback often limits the maximum usable gain of acoustic systems and degrades the overall system response. It is well known to be detrimental that the system stability and performance must be taken into account in system design. Most of the conventional methods for acoustic feedback cancellation in an acoustic system are based primarily on an adaptive filter with the least-mean-square (LMS) error algorithm. Unfortunately, convergence speed is often limited when a sound source or a filtering plant is varied, because the learning process of the adaptive algorithm fails to respond fast enough to changing operational conditions. This report proposes a variable step-size affine-projection algorithm (VSS APA) for acoustic feedback cancellation in audio systems. The proposed adaptive filter is based on the filtering affine-projection algorithm with variable step-size for improving convergence speed in acoustic feedback cancellation. A performance evaluation and simulation comparison has been conducted to compare the proposed algorithm and various traditional adaptive filtering algorithms.

Ideally, signals in statistical key exchange should be band-limited to prevent leak. During digitization, however, the broadband quantization noise gets incorporated. In this study, the security risk posed by the quantization noise is analyzed. The Savitzky–Golay filtering technique is used to reduce the quantization noise. Although the noise can be reduced, it cannot be eliminated. Legitimate users must take evasive measures. The adversary can perform exhaustive search. Once how signals are processed is identified, the security can be breached.

This paper establishes novel criteria for the induced $l_{\infty }$ stability to avoid overflow oscillations in fixed-point digital filters with generalized overflow non-linearities and external noise. The proposed linear matrix inequality (LMI)-based criteria ensure exponential stability as well as confirm reduction in the influence of external noise. The generalized overflow non-linearities which are considered for analysis commonly occur in practice, viz. saturation, zeroing, two's complement, and triangular. The presented approach unifies a string of existing results which are derived by considering saturation non-linearities and external interference. Simulation examples are shown to validate the usefulness of the proposed approach.

This paper proposes a novel criterion for suppressing the $H_{\infty }$ overflow oscillations in fixed point state-space digital filters employing saturation nonlinearities and external interference. The proposed criterion can be used to ensure the exponential stability (ES) and diminish the external interference effects to an $H_{\infty }$ norm constraint. An example is given to exemplify the utility of the obtained results.

This paper deals with the problem of mixed $H_{\infty }$ and passivity performance analysis of digital filters subject to Markovian jumping parameters, external disturbances, time delays and bounds of the nonlinearity functions. By employing Lyapunov theory and matrix decomposition technique, a novel sufficient condition is established. The proposed criterion ensures that the underlying system is stochastically stable and satisfies a mixed $H_{\infty }$ and passivity performance index simultaneously. The obtained criterion can also be employed to solve the $H_{\infty }$ problem or the passivity problem in a unified framework. Moreover, the problem is formulated to obtain optimal mixed $H_{\infty }$ and passivity performance index of the interfered digital filters. The effectiveness and superiority of our proposed results are illustrated by three examples.

Frequency weighting networks are a critical component of a sound level meter (SLM), and their error characteristics directly determine the performances of SLM. For reducing the high-frequency error of the $A∕C$ frequency-weighting filters with the bilinear transformation method (BTM), a design method for $A∕C$ frequency-weighting filters based on neural computing method (NCM) is proposed. A detailed algorithm for solving the filter coefficients is provided, and the amplitude-frequency characteristics of the $A∕C$ frequency-weighting filters with BTM and NCM are compared in detail. The experimental results show that the amplitude-frequency characteristics of the $A∕C$ frequency-weighting filters in SLM with NCM are significantly better than those of BTM. The filter meets the requirements of the first class SLM defined by IEC61672, which demonstrates the effectiveness of this proposed method.

This paper presents a preliminary investigation on nonlinear wave propagation in multistory buildings. The proposed one-dimensional nonlinear wave equation solver in time domain is demonstrated to yield favorable results as conventional modal analysis method, yet it carries a most desirable feature that wave arrival time delay and higher mode contribution can be included properly. The proposed method will become superior when more sophisticated digital filter design is launched in future studies.

This article explains an approach to further enhance detection of straight lines and arcs, using subpixel-precision methods, based on the contour points approach by Schumann¹ using polynomial based filtering methods.

To improve the defect of finite impulse response (FIR) digital filter design, like complicated theory and calculation, the Particle Swarm Optimization (PSO) algorithm is introduced to solve the multi-parameter optimization problem. And catfish effect in nature and chaos are also taken into consideration separately to improve PSO algorithm, where catfish effect is used to improve the population search strategy and chaos is to transform the individual evolutionary strategy. Due to this, the two ways are integrated together then a more stable passband, greater stop-band attenuation and narrower transition band width than other three methods is obtained.