In this paper we present a concept of a camera based on a line scanning sensor. The core of the camera is a proprietary broadband MEMS-based row imaging sensor and a powerful multiprocessor chip. We present here the architecture of the camera as well as selected subblocks of the system.

This paper presents a robust evaluation of partial subfilters in a cascade representation of equiripple narrow band-pass FIR filters. The presented procedure is based on an approximating equiripple polynomial. We demonstrate its functionality and robustness.

This paper presents an improved design of optimal equiripple notch FIR filters. In contrast to previous achievements, the notch FIR filters are additionally specified by the stop-band properties which are reflected by a novel degree equation. The degree equation relates the filter specification and the lowest possible filter length. An enhanced filter design is presented. Two examples demonstrate its functionality and robustness.

A simple yet powerful procedure for an echo attenuation in signals is introduced. The presented method involves no external reference signal. It is based on comb FIR filtering. To the advantages of the described method belong the simplicity and performance which are beneficial in real time implementations. For illustration, a simulation of the procedure is included. The efficiency of the presented method is demonstrated by a real time implementation on a digital signal processor.

This brief introduces a novel closed-form method for a robust evaluation of partial impulse responses in a cascade structure of narrow equiripple (ER) bandpass finite-impulse-response filters. The approach presented here is based on a generating polynomial for an ER approximation. The importance of the method is seen in its inherent robustness, which significantly outperforms the existing numerical approach. Consequently, much longer filters can be represented in their cascade form. Two examples demonstrate the functionality, performance, and usefulness of the method, as well as its superiority in terms of robustness over the existing approach.

This brief presents the degree equation of an equiripple narrow bandpass finite-impulse response filter, which follows from the filter approximation rather than from empirical data. The degree equation relates the filter specification and the corresponding filter degree. It completes the closed-form filter design, and it is useful in numerical design procedures as well. Two examples demonstrate its functionality and usefulness.

A closed form design of narrowest possible band-pass finite impulse response filters is presented. The underlaying generating polynomial has equiripple behaviour in two separate intervals. A novel degree equation is introduced. One example demonstrates the functionality of the design procedure.

This paper presents a real-time implementation of a Morse code decoder based on the TMS320C6748 digital signal processor. The hardware platform is the TMS320C6748 DSP development kit LCDK. The decoder processes a noisy audio signal, detects Morse signals, selects the strongest one, extracts its dots and dashes, decodes the Morse code and displays the decoded symbols on the VGA monitor in real-time.

A closed form design of nearly equiripple low-pass finite impulse response filters is introduced. The approximation is based on the generating polynomial, which is related to the iso-extremal polynomials. Formulas for the evaluation of the impulse response of the filter has been developed. The practical design procedure based on this approximation is presented. One example is included.

The digital subscriber lines are still base of access network to provide Internet connectivity to the public. The transmission capacity is continually increased and performance of digital subscriber lines must be estimated for network provider use. This paper describes two ways for correction of digital subscriber line modelling. The resulting bitrates are calculated with respect to the length of the line and the number of disturbance sources. The article brings new ideas for simulations and calculations of transmission function for line with inhomogeneity and calculations new value of code gain.

Filter sets based on narrow band-pass finite impulse response filters are introduced. The presented filter sets consist of inner and lateral filters. The inner filters are equiripple narrow band-pass finite impulse response filters based on isoextremal polynomials. They are complemented by the lateral narrow low-pass and high-pass finite impulse response filters. The filter sets presented here enable the perfect decomposition of the input signal.

The base of electric circuit analysis is often set up on numerical methods. When taking a look into the principle of any numerical method, a matrix calculus always rises out. The larger the problem is, the larger is the size of the matrix, representing the logical structure of the solved problem [1]. The problem (especially from the mathematical point of view) is in acquiring the inverse matrix, which can be solved with some of a large scale of available numerical (or analytical; based on the size of matrixes) methods. This paper shows another possible approach to obtaining the inverse matrix. The graph theory has already been successfully used to invert the matrixes; although, the idea of evaluating the alternating paths, presented by authors, is completely new and had not been used before. The utilization of the designed procedure is demonstrated on a part of an analogue signal processing filter. This technology gives a new tool to software designers and could speed up the numerical solutions; or, in case of impossible analytical and/or diverge numerical technique can lead to a successful and reasonable solution.

The unique project based on cooperation of universities and primary or secondary schools is transformed to Czech Republic. The project, realized at Czech Technical University in Prague, Faculty of Electrical Engineering is focused on support of technical education and knowledge between primary and secondary schools' students (and teachers). The main idea is based on “Engineer Olympics” realized at Faculty of Mechanical Engineering, STU in Bratislava [1], and “Technical Olympics — Science-Research-Development” realized at Faculty of Mechatronics in Trencin, Slovakia (both projects started at 2008), and project “Science-Research-Development in a View of Children”. The paper describes the basic ideas of the project, its three different steps as well as its advantages in comparison to other similar activities. The important part of the paper is also the evaluation of previous four very successful seasons of the project, realized in Slovak Republic. There are almost 68000 pageviews [5] registered during the mentioned four years. The qualitative and quantitative statistics will be discussed and the authors' experiences, conclusions and advice will be presented.

Implementation of high performance linear phase FIR filters is presented. The filters are implemented on a high performance eight-core digital signal processor TMS320C6678. Selected aspects of the filter implementation are discussed.

The matrix calculus is very frequently used in electrical engineering to calculate the different parameters of (but not only) electric circuits. One of the most important points of mathematical background of this analysis is the calculation of inverse matrix. This can be realised by numerous well-known algebraic methods (unit/identity matrix, different algebraic supplements, etc.) or numerical techniques. Inverting of matrixes is a well-known method; nevertheless, idea of alternating paths (presented in the paper) is completely new and unpublished. The electric circuit can be represented in the form of a graph and this graph structure rewritten into the form of matrixes. The paper shows the inverse matrix acquisition without using any algebraic technique - directly from graph structure. The presented technology can further be used as an input for any numerical method used for circuit parameters analysis or on- and off-line simulation. Practically, this is based on the following steps: the electric circuit is transformed into the graph structure. A new graph structure is created using the presented algorithm. The matrix system describing this graph structure, obtained without use of any conventional technique, is the inverse matrix of the original graph. The proposed utilization of alternative paths essentially differs from work of other authors.

An advanced design of two-dimensional linear phase notch FIR filters is presented. The completely analytical design method is based on Zolotarev polynomials. A closed form solution of the design of two-dimensional notch FIR filters is presented. One example demonstrates the efficiency of the presented design procedure.

A two-dimensional pointing mirror for a laser beam deflection is presented. The mirror is primarily aimed for the pointing of a laser beam in the free space communication. The deflection of the mirror is accomplished by means of three voice coil actuators used in hard disk drives.

An advanced education in the real-time digital signal processing is presented. A practical training based on implementation of algorithms on digital signal processors is emphasized. Three laboratory tasks are presented. The long-year experience is summarized. Several recommendations are presented.

This paper pertains to speech and acoustic signal processing, particularly to the determination of echo path delay and operation of echo cancellers. To cancel long echoes, the number of weights in a conventional adaptive filter must be large. The length of the adaptive filter will directly affect both the degree of accuracy and the convergence speed of the adaptation process. We present a new adaptive structure which is capable to deal with multiple dispersive echo paths. An adaptive filter according to the present invention includes means for storing an impulse response in a memory, the impulse response being indicative of the characteristics of a transmission line. It also includes a delay estimator for detecting ranges of samples within the impulse response having relatively large distribution of echo energy. These ranges of samples are being indicative of echoes on the transmission line. An adaptive filter has a plurality of weighted taps, each of the weighted taps having an associated tap weight value. A tap allocation/control circuit establishes the tap weight values in response to said detecting means so that only taps within the regions of relatively large distributions of echo energy are turned on. Thus, the convergence speed and the degree of estimation in the adaptation process can be improved.

A bstract— In this paper, we propose a method for enhancing of speech corrupted by broadband noise. The method is based on the spectral subtraction technique. Performance of spectral subtraction, its limitations, artifacts introduced by it, and spectral subtraction modifications for eliminating these artifacts are discussed in the paper in details. To eliminate the musical noise, one of the artifacts introduced by conventional spectral subtraction, we propose to implement reduced varying scaling factor of spectral subtraction, with a following application of weighted function. Weighting function, used in the proposed algorithm, attenuates frequency spectrum components lying outside identified formants regions. The algorithm effects a substantial reduction of the musical noise without significantly distorting the speech. Listening tests were performed to determinate the subjective quality and intelligibility of speech enhanced by our method. Proposed noise reduction algorithm is compared to conventional spectral subtraction based on SNR improvement introduced by them. Spectrograms of speech, enhanced by the proposed algorithm and other modified spectral subtraction algorithms, which show the algorithms performance and degree of speech distortion, are also presented in the paper.