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An approach for improving of efficiency the operation of distributed control system of aviation engine based on wireless technology with high productivity and resolution of wave distributed surface has been presented. It can be applied for development of new principles of correct location the nodes, including the data processing equipment, the intellectual sensors, actuators, repeaters, central units in adaptive control strategies of aviation engine. The optimization method for processing information, using adaptive wavelet filters, as an optimal filter, that minimizes the average square of a common error for organizing the connection between wireless elements in the control systems of aviation gas turbine engine, has been considered, on the theoretical point of view. Also, the wavelets applications in the Wireless Distributed Automatic Control System (WDACS) for aviation engine, the requirements for the construction of its node, supported by the protocol stack, the scheme and the programs with a combination of connections the information exchange between elements have been considered, from the practical point of view. [ABSTRACT FROM AUTHOR]

In this paper, a method is proposed to reduce artifact effect for images. The proposed approach first decomposes an original image into four subband images. The powers of the four sub-images are calculated and compared with a power threshold accompanying entropy measure to decide that whether the sub-images are reserved or not. The pixels of the reserved sub-images are then compared with a pixel threshold accompanying different entropy measure in order to determine that the pixel is discarded or not. The experimental results of this approach in comparison with other methods are performed both subjectively and qualitatively and shown that the proposed algorithms could reconstruct the desired recovered performance subjectively and objectively. [ABSTRACT FROM AUTHOR]

A multiplierless architecture based on algebraic integer representation for computing the Daubechies 4-tap wavelet transform for 1-D/2-D signal processing is proposed. This architecture improves on previous designs in a sense that it minimizes the number of parallel 2-input adder circuits. The algorithm was achieved using numerical optimization based o exhaustive search over the algebraic integer representation. The proposed architecture furnishes exact computation up to the final reconstruction step, which is the operation that maps the exactly computed filtered results from algebraic integer representation to fixed-point. Compared to Madishetty et al. (IEEE Trans Circuits Syst I (Accepted, In Press), ), this architecture shows a reduction of $$10\cdot n-3$$ adder circuits, where $$n$$ is the number of wavelet decomposition levels. Standard $$512\times 512$$ images Mandrill, Lena, and Cameraman were submitted to digital realizations of both proposed algebraic integer based as well as fixed-point schemes, leading to quantifiable comparisons. The design is physically implemented for a 4-level 2-D decomposition using a Xilinx Virtex-6 vcx240t-1ff1156 FPGA device operating at up to a maximum clock frequency of 263.15 MHz. The FPGA implementation is tested using hardware co-simulation using an ML605 board with clock of 100 MHz. A 45 nm CMOS synthesis shows improved clock frequency of better than 500 MHz for a supply voltage of 1.1 V. [ABSTRACT FROM AUTHOR]

This article presents a method for increasing the performance of the progressive coding algorithms for the subbands of images, by representing the coefficients with a code that reduces the truncation error. [ABSTRACT FROM AUTHOR]

In this paper, a computationally efficient iterative algorithm is presented for the design of multi-channel nearly perfect reconstruction nonuniform filter bank using the modified window functions such as Kaiser, Cosh and Exponential windows with exploiting a new perfect reconstruction condition of nonuniform filter banks instead of using complex objective functions. The cutoff frequency is optimized using linear optimization technique such that the magnitude response of a prototype filter at quadrature frequency is approximately equal to 0.707. The simulation results illustrate significant reduction in amplitude distortion, number of iteration and computation time as compared to earlier existing techniques. The proposed algorithm is simple, easy to implement, and linear in nature. When exploited for subband coding of electrocardiogram (ECG) signals, the proposed method yields good fidelity performance measuring parameters. [ABSTRACT FROM AUTHOR]

This paper presents an improved and efficient method for the design of a two-channel quadrature mirror filter (QMF) bank. In the proposed method, the filter bank design problem is formulated as a low-pass prototype filter design problem, whose responses in the passband and stopband are ideal and their filter coefficients value at quadrature frequency is 0.707. A new method is developed for the design of a low-pass prototype filter which minimizes the objective function by optimizing the filter taps weights using the Levenberg-Marquardt method. When compared with other existing algorithms, it significantly reduces peak reconstruction error ( PRE), error in passband, stopband and transition band. Several design examples are included to show the increased efficiency and the flexibility of the proposed method over existing methods. An application of this method is considered in the area of subband coding of the ultrasound images. [ABSTRACT FROM AUTHOR]

An improved and simplified approach is presented for design of nearly perfect reconstructed cosine-modulated (CM) filter banks with prescribed stopband attenuation and channel overlapping. The method employs Kaiser window technique to design the prototype filter for filter banks with the novelty of exploiting spline functions in the transition band of the ideal filter instead of using the conventional brick-wall filter based on linear optimization of filter coefficients such that their value at frequency ( ω = π/2 M) is 0.707. The simulation results illustrate the proposed method and its improvement over other existing methods in terms of amplitude distortion ( e ), number of iterations ( NOI), aliasing distortion ( e ) and computation time (CPU time). [ABSTRACT FROM AUTHOR]

Multiple description coding is one of the source coding techniques used to alleviate the problems of packet loss in network. The objective is to encode a source into two (or more) bitstreams supporting two quality levels of decoding: A high-quality reconstruction decoded from the two bitstreams together, and two lower ones, but still acceptable, decoded from either of the two bitstreams individually. Most of the earlier works concentrated on the trade-off between redundancy and average side distortion, which is insufficient to evaluate the system performance. In this paper, we discuss the difference in quality between the two side reconstructions, which is defined as the balance eccentric modulus of the coding system. A scenario to design balanced multiple description coding system is presented. Application to the popular first-order autoregressive model yields encouraging results. [ABSTRACT FROM AUTHOR]

The article focuses on a study which suggests an algorithm to design a two-channel linear phase quadrature mirror filter (QMF) bank using unconstrained optimization. It describes digital filter banks are essential parts of the multirate systems like in communication, multimedia, speech and biomedical signal processing. In the formulation of the design problem, a low pass prototype filter is created. The performance of the algorithm has been measured my amplitude distortion, peak reconstruction error and stopband attenuation.

In this paper, we present a novel audio coder using the discrete wavelet transform (DWT) and warped linear prediction (WLP). In contrast to conventional LP, WLP allows for the control of frequency resolution to closely match the response of the human auditory system. The structure of the system is similar to the transform coded excitation techniques used in wideband speech coding, where LP has been replaced with WLP, and the residual is analyzed by a wavelet filterbank designed to approximate the critical bands. The inherent shaping of the WLP synthesis filter, and a controlled bit allocation to the wavelet coefficients helps minimise the perceptually significant noise due to the quantization error in the residual. For monophonic signals sampled at 44.1 kHz, the coder achieves near transparent to transparent quality for a variety of speech and music signals at an average bitrate of about 64 kb/s. Tests also show that the coder (in its initial implementation) delivers superior quality to the MPEG layer III and comparable quality to the MPEG2-AAC codec when operating at the same bitrate. [ABSTRACT FROM AUTHOR]

The conventional two-dimensional wavelet transform used in existing image coders is usually performed through one-dimensional (1-D) filtering in the vertical and horizontal directions, which cannot efficiently represent edges and lines in images. The curved wavelet transform presented in this paper is carried out by applying 1-D filters along curves, rather than being restricted to vertical and horizontal straight lines. The curves are determined based on image content and are usually parallel to edges and lines in the image to be coded. The pixels along these curves can be well represented by a small number of wavelet coefficients. The curved wavelet transform is used to construct a new image coder. The code-stream syntax of the new coder is the same as that of JPEG2000, except that a new marker segment is added to the tile headers. Results of image coding and subjective quality assessment show that the new image coder performs better than, or as well as, JPEG2000. It is particularly efficient for images that contain sharp edges and can provide a PSNR gain of up to 1.67 dB for natural images compared with JPEG2000. [ABSTRACT FROM AUTHOR]

JPEG 2000 is a new image coding system that delivers superior compression performance and provides many advanced features in scalability, flexibility, and system functionalities that outperform all previous standards. It brings exciting possibilities to many imaging applications such as the Internet, wireless, security, and digital cinema. This paper gives an overview of this triumph in innovations and teamwork. It gives brief introductions to the four new parts that are under development by the JPEG committee. It attempts to summarize the lessons learned from the JPEG 2000 development experience and draw some conclusions on the success factors of this important standard. [ABSTRACT FROM AUTHOR]

One of the purposes of this article is to give a general audience sufficient background into the details and techniques of wavelet coding to better understand the JPEG 2000 standard. The focus is on the fundamental principles of wavelet coding and not the actual standard itself. Some of the confusing design choices made in wavelet coders are explained. There are two types of filter choices: orthogonal and biorthogonal. Orthogonal filters have the property that there are energy or norm preserving. Nevertheless, modern wavelet coders use biorthogonal filters which do not preserve energy. Reasons for these specific design choices are explained. Another purpose of this article is to compare and contrast “early” wavelet coding with “modern” wavelet coding. This article compares the techniques of the modern wavelet coders to the subband coding techniques so that the reader can appreciate how different modern wavelet coding is from early wavelet coding. It discusses basic properties of the wavelet transform which are pertinent to image compression. It builds on the background material in generic transform coding given, shows that boundary effects motivate the use of biorthogonal wavelets, and introduces the symmetric wavelet transform. Subband coding or “early” wavelet coding method is discussed followed by an explanation of the EZW coding algorithm. Other modern wavelet coders that extend the ideas found in the EZW algorithm are also described [ABSTRACT FROM PUBLISHER]

In many applications of filter banks, a low-delay filter bank in which the system delay is small is desirable. In this paper, two new design methods are proposed for low-delay perfect-reconstruction filter banks. First, a design method is described for a two-channel perfect-reconstruction filter bank suitable for subband coding. In this method, a high-pass filter for perfect reconstruction is designed for an arbitrary low-pass filter. Next, a design method is described for an oversampling perfect-reconstruction DFT filter bank suitable for subband adaptive filter. In this method, a prototype filter for a perfect-reconstruction synthesizer is designed for the prototype filter for an arbitrary analyzer. The effectiveness of the present method is demonstrated by design examples. © 2001 Scripta Technica, Electron Comm Jpn Pt 2, 84(6): 30–39, 2001 [ABSTRACT FROM AUTHOR]

This paper presents an image restoration model based on the implicit function theorem and edge-preserving regularization. We then apply the model on the subband-coded images using the artificial neural network. The edge information is extracted from the source image as a priori nowledge to recover the details and reduce the ringing artifact of the subband-coded image. The multilayer perceptron model is employed to implement the restoration process. The main merit of the presented approach is that the neural network model is massively parallel with strong robustness for the transmission noise and parameter or structure perturbation, and it can be realized by VLSI technologies for real-time applications. To evaluate the performance of the proposed approach, a comparative study with the set partitioning in hierarchical tree (SPIHT) has been made by using a set of gray-scale digital images. The experimental results showed that the proposed approach could result in compatible performances compared with SPIHT on both objective and subjective quality for lower compression ratio subband coded image. [ABSTRACT FROM AUTHOR]

A lossless coding method that can restore digital images without any distortion is needed in areas that rely on high-quality images such as medical and broadcast applications. Lossless subband coding can obtain a high compression performance and enables progressive image transmission, and hence is known to be an effective lossless coding method. In the conventional two-band lossless subband coder, there are restrictions in the filter tap length, filter coefficients, and type of filter bank such as separable or nonseparable. Therefore, it is difficult to design the subband coder flexibly in response to special needs such as to increase the compression efficiency for a specific image, or to reduce the distortion in the progressive image transmission. In this paper, a two-band lossless subband coder is proposed that has several degrees of freedom in selection of the filter tap length, filter coefficients, and type of filter bank. These degrees of freedom are used for improvement of compression efficiency for a specific image. The filter coefficients are determined by the nonlinear programming. The proposed method is a general form containing the conventional method. For several test images, a compression efficiency higher than that in the conventional two-band lossless subband coder is confirmed. © 1999 Scripta Technica. Electron Comm Jpn Pt 3, 83(4): 103–114, 2000. [ABSTRACT FROM AUTHOR]

Copyright of Annals of Telecommunications is the property of Springer Nature and its content may not be copied or emailed to multiple sites or posted to a listserv without the copyright holder's express written permission. However, users may print, download, or email articles for individual use. This abstract may be abridged. No warranty is given about the accuracy of the copy. Users should refer to the original published version of the material for the full abstract. (Copyright applies to all Abstracts.)

Filter banks are usually designed so that the perfect reconstruction condition is satisfied. This condition, however, is not always satisfied in actual applications. In such a case, the filter bank operates as a time-varying system, and checkerboard distortion is produced by the time-varying effect. This paper discusses the checkerboard effect in a two-channel filter bank, including the effect of finite word length implementation and the case in which the perfect reconstruction condition is not satisfied. The general condition for avoiding the checkerboard effect is formulated. A design procedure for a paraunitary QMF bank satisfying that condition, as well as an implementation by the lattice configuration, are shown. It is furthermore shown that the checkerboard effect can be minimized in a finite word length implementation by employing the proposed lattice configuration and adding a correction to the rotation angle. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(11): 74–85, 1999 [ABSTRACT FROM AUTHOR]

This paper points out problems in the reversible subband coding, together with the means to overcome the problems. It is shown for the reversible subband coding that there exists a redundancy between the subband signals, and that the reduction of the information is less in the subband decomposition of the high-band signal. In the proposed system, subband decomposition is not applied to the high-band signal, but the high-band signal is decomposed based on the similarity between subband signals. More precisely, the already reconstructed image with low resolution is decomposed according to the magnitude of the pixel value gradient, and the high-band signal is correspondingly decomposed. By such a decomposition method, it is made possible to transmit and display the signals from more important components. Then, it is expected that the compression efficiency is improved and that the psychological burden in successive image display can be reduced. It is shown by simulation that the compression efficiency is improved in lossless reconstruction and in progressive image reconstruction. © 1997 Scripta Technica, Inc. Electron Comm Jpn Pt 3, 80(9): 21–28, 1997 [ABSTRACT FROM AUTHOR]

Although the discrete cosine transform (DCT) and wavelet transform have been used as effective techniques for reducing the redundancy of image waveforms, they have the problems of high complexity and causing block distortion. In this paper, a Haar wavelet transform with interband prediction is proposed that permits high-speed processing and which is suitable for coding as a subband decomposition technique since it generates little block distortion in the decoded images. The interband prediction method of this proposed transform method uses the derivative in the decoded images. The interband prediction method of this proposed transform method uses the derivative of the low-band waveform to predict the high-band waveform each time a band is split into two subbands. This results in a transform process that gives a prediction residual signal with lower entropy. During the inverse transform, the prediction coefficients obtained from the forward transform are used to predict the high-frequency waveforms and then, b adding the prediction residual, the original high frequency waveforms are reconstructed. Because the high-frequency waveforms are generated that smoothly interpolate the up-sampled, low-frequency waveforms, any block effects from this prediction technique are difficult to see in the decoded image. A detailed analysis of the proposed interband prediction process is performed by interpreting the proposed transform as a subband decomposition process that is based on the use of a symmetric short-kerner filter (SSKF) filter bank; then experiments are conducted that demonstrate its performance as a subband decomposition for coding of real image data and the possibility of progressive build-up of the decoded images. Results show that this transform method is effective for coding image data. [ABSTRACT FROM AUTHOR]

Short tap filters reduce the computational complexity of filtering processes and also reduce image degradations associated with ringing effects near edges in image subband coding. When the number of taps is reduced, however, coding performance generally deteriorates. In this paper, short tap filters that satisfy isolated perfect reconstruction constraints are designed so as to improve coding performance. For the sake of comparisons, linear phase and orthogonal filters are also designed. Filters with coefficients determined by the values 2n are designed to reduce further the computational burden of the filter processing. Assuming that the input can be modeled by an AR(1) process, the set of parameters which determine the tap coefficients are selected to maximize the coding gain. The relationship between signal-to-noise ratio and entropy is determined to evaluate coding performance. The superior coding performance of the perfect reconstruction filters shows no degradation when subjected to an integer coefficient constraint. Also, filters with coefficients determined by the values 2n and excellent coding performance are obtained. [ABSTRACT FROM AUTHOR]

By decreasing the decimation ratio to one less than the number of channels, a perfect-reconstruction quadrature mirror filter (QMF) system can be constructed with the following innovative features: (1) the band division, which in the past has been uniform, can be made nonuniform; (2) the equivalent delay, which has usually been as large as the order of the dividing filter, can now be reduced to a size equal to the decimation ratio minus 1; (3) the number of channels, which has been restricted to a power of 2, can now be arbitrary. In addition, the traditional method employs spectral decomposition, while the proposed method employs Euclidean division. Euclidean structure (the realization of Euclidean division on a network) and lattice structure are considered as methods of constructing a filter that does not produce an aliasing distortion for the finite word length. [ABSTRACT FROM AUTHOR]

A new variable bit rate layered coding scheme based on subband coding is given in this paper. It can make full use of the merits of ATM transmission, as well as compensation for the quality degradation caused by cell loss, which is an inherent problem of ATM networks, and thus a stable, high quality video transmission is achieved. This paper presents the coding scheme, cell construction and simulation results. [ABSTRACT FROM AUTHOR]

In this work, we propose a coding technique that is based on the generalized block prediction of the multiresolution subband decomposition of motion compensated difference image frames. A segmentation mask is used to distinguish between the regions where motion compensation was effective and those regions where the motion model did not succeed. The difference image is decomposed into a multiresolution pyramid of subbands where the highest resolution subbands are divided into two regions, based on the information given by the segmentation mask. Only the coefficients of the regions corresponding to the motion model failure are considered in the highest resolution subbands. The remaining coefficients are coded using a multiresolution vector quantization scheme that exploits inter-band non-linear redundancy. In particular, blocks in one subimage are predicted from blocks of the adjacent lower resolution subimage with the same orientation. This set of blocks plays the role of a codebook built from coefficients inside the subband decomposition itself. Whenever the inter-band prediction does not give satisfactory results with respect to a target quality, the block coefficients are quantized using a lattice vector quantizer for a Laplacian source. [ABSTRACT FROM AUTHOR]