The designing of transmultiplexer systems relies on determining filters for the transmitter and receiver sides of multicarrier communication system. The perfect reconstruction conditions lead to the bilinear equations for FIR filter coefficients. Generally there is no way of finding all possible solutions. This paper describes methods of finding a large family of solutions. Particular attention is devoted to obtaining algorithms useful in fixed-point arithmetic needed to design the integer filters. As a result, the systems perform perfect reconstruction of signals. Additionally, a simple method is presented to transform any transmultiplexer into an unlimited number of different transmultiplexers. Finally, two examples of integer filters that meet perfect reconstruction conditions are shown. The first illustrates a FIR filter which does not require multiplications. The frequency properties of filters and signals are discussed for the second example.

Commonly known DC-AC switching converters are commonly used in compensator branches. One example of this is a static synchronous compensator (STATCOM). It consists of a voltage source converter (VSC) and acts as an inverter with a capacitor as a DC power source. These compensators use the PWM switching scheme or space vector modulation (SVM) method. Both methods require the desired signal to be generated. In some cases, as during the synthesis of self-excited systems or active energy-compensators, it is necessary to perform the desired branch immittance, e.g. negative capacitance, inductance, resistance or irrational impedance. In such cases, it is necessary to control the universal branch on the basis of a formula. This article presents the implementation method for the convolutional type impedance operators.

In this paper the way of modeling phenomena occurring during the voltage and current waves passing through a point connection of two lines, with different wave impedance operators, is presented. This connection point is called „the wave transformer”. The analyzes and the resulting formulas concern not the frequency domain, but the time domain. The appropriate transition matrices of waves through the wave transformer are defined. This matrices are the convolution integral-derivative operators of fractional order (the digital filters). For a lossless line the wave transition matrices through the wave transformer become number type instead of operator type. All matrix multiplications occurring in the formulas should be understood in convolution way.

Many studies have been developed aiming to improve digital filters realizations, recurring to intricate structures and analyzing probabilistically the error's behavior. The work presented in this paper analyzes the feasibility of fixed-point implementation of classical infinite impulse response notch filters: Butterworth, Chebyshev I and II, and elliptic. To scrutinize the deformations suffered for distinct design specifications, it is assessed: the effect of the quality factor and normalized cut-off frequency, in the number of significant bits necessary to represent the filter's coefficients. The implications brought to FPGA implementation are also verified. The work focuses especially on the implementation of power line notch filters used to improve the signal-to-noise ratio in biomedical signals. The results obtained, when quantizing the digital notch filters, show that by applying second-order sections decomposition, low-order digital filters may be designed using only part of double precision capabilities. High-order notch filters with harsh design constraints are implementable using double precision, but only in second-order sections. Thus, it is revealed that to optimize computation time in real-time applications, an optimal digital notch filter implementation platform should have variable arithmetic precision. Considering these implementation constraints, utmost operation performance is finally estimated when implementing digital notch filters in Xilinx Virtex-5 field-programmable gate arrays. The influence of several design specifications, e.g. type, and order, in the filter's behavior was evaluated, namely in regard to order, type, input and coefficient number of bits, quality factor and cut-off frequency. Finally the implications and potential applications of such results are discussed.

In the paper an application of differential evolution algorithm to design digital filters with non-standard amplitude characteristics is presented. Three filters with characteristics: linearly growing, linearly falling, and non-linearly growing are designed with the use of the proposed method. The digital filters obtained using this method are stable, and their amplitude characteristics fulfill all design assumptions.

Baseflow is the primary source of water for irrigation and other water needs during prolonged dry periods; accurate and rapid estimation of baseflow is therefore crucial for water resource allocation. This research aims to estimate baseflow contribution during dry periods in three small watersheds in East Java: Surabaya-Perning (114 km2), Lamong-Simoanggrok (235 km2), and Bangsal-Kedunguneng (26 km2). Six recursive digital filters (RDFs) algorithms are explored using a procedure consisting of calibration, validation, evaluation and interpretation. In this study, the period of July to September is considered as the peak of the dry season. Moreover, data for the period 1996 to 2005 is used to calibrate the algorithms. By yearly averaging, values are obtained for the parameters and then used to test performance during the validation period from 2006 to 2015. Statistical analysis, flow duration curves and hydrographs are used to evaluate and compare the performance of each algorithm. The results show that all the filters explored can be applied to estimate baseflow in the region. However, the Lyne–Hollick (with RMSE = 0.022, 0.125, 0.010 and R2 = 0.951, 0.968, 0.712) and exponentially weighted moving average or EWMA (with RMSE = 0.022, 0.124, 0.009 and R2 = 0.957, 0.968, 0.891) for the three sub-watersheds versions give the best performance.

The aim of the work is to present the method for designing sparse FIR filters with very low group delay and approximately linear-phase in the passband. Significant reduction of the group delay, e.g. several times in relation to the linear phase filter, may cause the occurrence of undesirable overshoot in the magnitude frequency response. The method proposed in this work consists of two stages. In the first stage, FIR filter with low group delay is designed using minimax constrained optimization that provides overshoot elimination. In the second stage, the same process is applied iteratively to reach sparse solution. Design examples demonstrate the effectiveness of the proposed method.