|Digital Signal Processing||
Weighted Chebyshev approximation in design of pulse-shaping FIR filters for digital communication systems
This paper discusses the problem of design of a pair of matched pulse shaping linear-phase FIR filters. The design involves obtaining a predetermined attenuation of the magnitude response in the stopband and a minimum level of ISI. The known design method based on the weighted Chebyshev approximation using the Remez algorithm with additional control of the transition band at one point and weight selection for the magnitude response level in the stopband is compared with several alternative approaches namely the SRRC function, nonlinear programming, convex optimization and semi-analytical procedure. Examples of pulse-shaping filters taken from the literature show that this method does not always result in repetition or improvement of known solutions. However, the modification proposed in the paper, related to the addition of several more conditions for the control of the transition band, can contribute to a significant improvement in these solutions both for stopband attenuation at the same ISI value and for each of these parameters.
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Keywords: linear system identification, adaptive filters, cascaded adaptive filter, simplified adaptive filter, Recursive Least Squares, RLS algorithm, Matrix Inversion Lemma, correlation matrix, matrix diagonalization.
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The error in determining the location by the proposed algorithm compared with the existing hyperbolic method at 8178 points in the Asia-Pacific region. The data obtained made it possible to simulate and evaluate the working zones of two jointly used chains of pulse-phase radio navigation systems according to both algorithms. It shown that the use of the iterative position search method significantly expands the working area of chains when they used together.
In addition, in most areas, the position determination error is significantly lower than in the hyperbolic method. A method introduced for quantitative assessment of the effectiveness of the iterative position search method in comparison with the hyperbolic method.
Modeling the working area of several chains allows you to improve the efficiency of determining the deployment locations and formats of new systems.
A quantitative analysis of the effectiveness of the proposed method allows us to conclude that when using the iterative position search algorithm, the size of the zone in which the position error is below 20 meters is approximately 11 times larger than the zone in which the hyperbolic method provides the same error. For an error below 40 meters, the size of the working area increases by a factor of 2.86. For an error below 60 meters, 2.7 times.
The iterative position search algorithm is able to work with signals from two or more radio navigation circuits. The use of the results of the study will make it possible to determine the optimal deployment sites and formats for the operation of new combined radio navigation systems.
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Classification of aerial targets based on a system with a random jump-like structure using information from neural network classifiers
Recognition of small UAVs is not an easy task due to the similarity of the radar characteristics of such targets and the parameters of their movement both among themselves.
The recognition of targets can be carried out on the basis of the following features: the nature of the Doppler portrait of the reflected signal due to the peculiarities of the rotation of the propellers of a particular type of aircraft or the flapping of the wings of birds; radar image of the target formed by methods of inverse aperture synthesis; trajectory signs of the flight of the target, etc.
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Based on the introduced criterion, a two-stage optimization problem has been solved. At the first stage, the optimal RF vector is determined, at the second stage - a multichannel filter (MF) of coherent accumulation. Optimization results are presented depending on the correlation properties of the interference and a comparison is made with the efficiency of optimal processing.
The dependences of the optimal RF order on the magnitude of the dynamic range of interference in relation to the level of intrinsic noise are obtained. having a directly proportional character. The conditions are established under which a system of a fixed structure, the scheme of which is given, is achieved close to potential efficiency. The conditions for the use of tunable structure systems are considered, in which it is possible to approach the potential efficiency when changing the interference parameters in a relatively wide range only when optimizing the RF order by appropriate restructuring of the structure.
A method for choosing the RF and MF orders is proposed, based on the relationship of the optimal RF order with the increment of the interference transmission coefficient at the RF output when its order changes. As a result of the analysis of the dependences of the increments of the passage coefficient, the condition for choosing the optimal order of the RF is established. A block diagram of the adaptive processing system of the transferred structure is given.
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The AMR-WB speech codec is based on the algebraic code excited linear prediction technology (ACELP). Two frequency bands, 50 6400 Hz and 6400 7000 Hz, are coded separately.
The input signal of the lower frequency band is pre-processed using a high-pass filter and a pre-emphasis filter. Linear Prediction (LP) analysis is performed on each frame. The set of LP parameters is converted to immittance spectrum pairs and vector quantized using split-multistage vector quantization. The speech frame is divided into four subframes of 5 ms each. The adaptive and fixed codebook parameters and pitch lag are transmitted every subframe. The bit allocation of the codec at different bit rates is shown.
The higher frequency band is reconstructed in the decoder using the parameters of the
lower band and a random excitation. No information about the higher band is transmitted, except
in the 23,85 kb/s mode, where the higher band gain is transmitted. In other modes, the gain of the
higher band is adjusted relative to the lower band using voicing information. The spectrum of the
higher band is reconstructed by using a wideband LP filter generated from the lower band LP
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