Sampling Rate Alteration Research Articles

Transmultiplexer filter bank systems: a research overview

During last two decades, an extensive research has been carried out for the design and development of digital filter banks. These systems operate in two modes, i.e. either in analysis/synthesis mode or synthesis/analysis mode. First case corresponds to filter banks, in which input signal is decomposed into two or more signals. Second case corresponds to transmultiplexer systems, where several signals are transferred through a single channel. Transmultiplexer is a foremost module of multicarrier communication scheme, which is obtained by reversing the role of analysis and synthesis filters in the filter banks. They are often used in channel coding application in communication, and are also referred as a multi-input, multi-output system that uses sampling rate alteration and filtering to combine M signals for transmission across a channel, and then recovers the M input signals. This paper presents a comprehensive overview of transmultiplexer systems: analysis and design techniques of transmultiplexer systems. Finally, some concluding remarks about current and future research in transmultiplexer systems are presented.

A Multi-Level Robust and Perceptually Transparent Blind Audio Watermarking Scheme Using Wavelets

Abstract In this paper, a robust and perceptually transparent single-level and multi-level blind audio watermarking scheme using wavelets is proposed. A randomly generated binary sequence is used as a watermark, and wavelet function coding is used to embed the watermark sequence in audio signals. Multi-level watermarking is used to enhance payload capacity and can be used for a different level of security. The robustness of the scheme is evaluated by applying different attacks such as filtering, sampling rate alteration, compression, noise addition, amplitude scaling, and cropping. The simulation results obtained show that the proposed watermarking scheme is resilient to various attacks except cropping. Perceptual transparency of watermark is measured by using Perceptual Evaluation of Audio Quality (PEAQ) basic model of ITU-R (PEAQ ITU-R BS.1387) on Speech Quality Assessing Material (SQAM) given by European Broadcasting Union (EBU). Average Objective Difference Grade (ODG) measured for this method is -0.067 and -0.080 for single-level and multi-level watermarked audio signals, respectively. In the proposed single-level digital audio watermarking scheme, the payload capacity is increased by 19.05% as compared to the single-level Chirp-Based Digital Audio Watermarking (CB-DAWM) scheme.

Smart Front-End Signal Processing for Advanced Wireless Receivers

One of the key trends in the design of radio receivers and other wireless devices is to shift more and more of the transceiver functionalities to digital signal processing (DSP). At the same time, the demands on the remaining analog circuits are greatly increased, especially with lower power supplies and nanoscale technology effects such as variability. With the terminal users requesting high radio performance and data rates, and low power consumption on one hand, and terminal flexibility and reduced implementation costs on the other hand, the requirements for these remaining analog front-end stages become extremely challenging to meet. As a result, one interesting idea is to complement analog radio-frequency (RF) circuits with smart signal-processing algorithms to digitally enhanced RF circuits. In this paper, we focus on developing and demonstrating novel signal-processing techniques intended for the analog and digital front-ends of future low-power, flexible radios. One key aspect in the work is power-efficient digital front-end design with great flexibility for digital selectivity filtering and sample rate alteration. Another key ingredient is the analysis and mitigation of different analog RF impairments, with special emphasis on I/Q imbalance effects and second-order intermodulation (IM2) distortion in wideband multicarrier or multichannel radio receivers. The approach used in this work generally draws from the practical system performance specifications. Overall, our results clearly indicate that the proposed compensation techniques can be used to suppress I/Q imbalance and IM2 distortion effects in receiver front-end sections under realistic signaling assumptions. The adaptivity and flexibility offered by the overall digital front-end design greatly reduces the power consumption of the radio.

Efficient Implementation of Real and Complex Linear-Phase FIR and Minimum-Phase IIR Halfband Filters for Sample Rate Alteration

Efficient Implementation of Real and Complex Linear-Phase FIR and Minimum-Phase IIR Halfband Filters for Sample Rate Alteration

Multirate IIR filter structures for arbitrary bandwidths

This paper introduces new multirate infinite-length impulse-response (IIR) filter structures for arbitrary bandwidths. The overall filters have the same input and output sample rates. Multirate techniques are only used internally in order to improve the efficiency. The overall filters make use of an IIR filter for the actual filtering, periodic allpass filters for constructing complementary filters, and linear-phase finite-length impulse-response (FIR) filters for the sampling rate alterations. There are the following two main reasons for introducing the new filters: 1) they can be used to obtain so called high-speed IIR filters (i.e., filters able to operate at a high sample rate) with lower complexity than that achievable with single-rate high-speed filters and 2) they can be used to obtain approximately linear-phase filters with substantially lower complexity than that achievable with single-rate filters. The overall filters are designed by separately optimizing a number of linear-phase (possibly half-band) FIR filters, and one IIR filter being realizable as a parallel connection of two allpass filters. Design examples are included illustrating the usefulness of the proposed filters.

Projeto de banco de filtros para estruturas adaptativas em subbandas

Em filtragem adaptativa, freqüentemente são propostas novas estruturas e novos algoritmos que visam a acelerar a convergência e/ou dimimuir a complexidade computacional, principalmente em aplicações que requerem o uso de um número elevado de coeficientes adaptativos. Neste sentido estruturas de filtragem adaptativa em subbandas vêm ganhando destaque pela possibilidade de se fazer a filtragem e a adaptação em uma taxa de amostragem mais baixa do que a do sinal de entrada. Recentemente, novas estruturas adaptativas em subbandas capazes de modelar exatamente qualquer sistema de resposta ao impulso finita foram propostas. Neste trabalho, as propriedades de duas destas estruturas são investigadas: na primeira, composta de um banco de análise e de filtros esparsos, não há modificação da taxa de amostragem dos sinais, enquanto que na segunda, os sinais de saída do banco de análise são criticamente sub-amostrados. Procedimentos de otimização são propostos com o objetivo de selecionar os coeficientes dos bancos de filtros que resultem em uma redu cão do erro quadrático médio (MSE) e/ou na acelera cão da taxa de convergência dos algoritmos adaptativos.

Use of computers for learning basic concepts of sampling rate alteration

Use of computers for learning basic concepts of sampling rate alteration

Efficient rational sampling rate alteration using IIR filters

The problem of changing the sampling rate of a signal by a rational factor of L/M is discussed. It is shown that infinite impulse response (IIR) filters can be efficiently implemented using the polyphase decomposition. The computational cost of the recursive and nonrecursive parts of the interpolation filter are considered separately, and a gain in efficiency of a factor of LM/(L+M-1) is achieved for the recursive part. This gain is only significant when both L and M are larger than one.

Use of computers for learning basic concepts of sampling rate alteration

This paper presents a computer demonstration created with the idea of providing a better understanding of basic concepts of sampling alteration. The software package has been developed to complement the theory of multirate digital signal processing, taught at the Department of Electronics, INAOE, at graduate level. The package was implemented in MATLAB v. 5.2 and uses a graphical user interface or GUI tool available in MATLAB. The main demonstrations include discrete signal representation, decreasing the sampling rate, increasing the sampling rate, and the cascade of decimation and interpolation. © 2000 John Wiley & Sons, Inc. Comput Appl Eng Educ 8: 191–195, 2000

Use of computers for learning basic concepts of sampling rate alteration

Use of computers for learning basic concepts of sampling rate alteration

Predictive interpolation and decimation of narrow-band signals

A digital filter design and optimization method for sampling rate alteration of narrow-band signals is presented. The resulting predictive finite-duration impulse-response filters are used to remove the aliasing or imaging frequency components when the sampling rate is changed. The sampling rate can be changed either by an integer factor or a rational factor. A least-squares-based generalization is developed for optimal wide-band noise attenuation.

Efficient deformable filter banks

This article describes efficient schemes for the computation of a large number of differently scaled/oriented filtered versions of an image. We generalize the well-known steerable/scalable (deformable) filter bank structure by imposing X-Y separability on the basis filters. The resulting systems, designed by an iterative projections technique, achieve substantial reduction of the computational cost. To reduce the memory requirement, we adopt a multirate implementation. Due to the inner sampling rate alteration, the resulting structure is not shift invariant. We introduce a design criterion for multirate deformable structures that jointly controls the approximation error and the shift variance.

Adaptive FIR filter structure based on the generalized subband decomposition of FIR filters

An adaptive structure based on a generalized structural subband decomposition of FIR (finite-impulse-response) filters is presented. The proposed structure implements an adaptive FIR filter of length N as a parallel connection of L branches, with each branch composed of a cascade of a fixed interpolator and a sparse adaptive subfilter containing at least L nonzero coefficients. There is no sampling rate alteration in this structure, and therefore no problems with aliasing occur. The interpolators are implemented by a computationally efficient transform (e.g., DCT, DFT). The proposed structure presents superior convergence performance for colored input signals when compared to the conventional direct-form LMS (least-mean-square) structure, with a very small increase in the number of operations. The advantages of using the subband structure in the adaptive line enhancer, acoustic echo canceller, and channel equalizer applications are shown through computer simulations. >

A generalized structural subband decomposition of FIR filters and its application in efficient FIR filter design and implementation

It is shown that any FIR (finite-impulse-response) transfer function can be realized as a parallel connection of interpolated FIR (IFIR) sections consisting of a cascade of a subfilter with a sparse impulse response and an FIR interpolator which fills in the missing samples of the corresponding subfilter. The set of interpolators can be realized as a subband filter bank, and as a result, each IFIR section contributes to the overall frequency response essentially within a given band of frequencies. The proposed decomposition can be considered as a generalization of the polyphase decomposition. It results in a computationally efficient structure in sampling rate alteration applications and also leads to faster FIR filter design algorithms. Examples are included to demonstrate the improvements in the speed of least-squares and minimax design algorithms based on the suggested structure. >

The role of integer matrices in multidimensional multirate systems

Various theoretical issues in multidimensional (m-D) multirate signal processing are formulated and solved. In the problems considered, the decimation matrix and the expansion matrix are nondiagonal, so that extensions of 1-D results are nontrivial. The m-D polyphase implementation technique for rational sampling rate alterations, the perfect reconstruction properties for the m-D delay-chain systems, and the periodicity matrices of decimated m-D signals (both deterministic and statistical) are treated. The discussions are based on several key properties of integer matrices, including greatest common divisors and least common multiples. These properties are reviewed.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Time domain analysis and design of multirate wave digital filters

AbstractA generalized time domain approach is described for the analysis and design of wave digital filters capable of performing sampling rate alterations whether integer or non‐integer. It is based on sequential computation of the filter's system response; then this discrete response is reduced to a rational function form. the proposed approach is characterized by its simplicity as well as its ability to analyse and design multistage, multirate wave digital structures. Illustrative examples are also given.

Compression of Two-Dimensional Band-Limited Signals Using Sub-Sampling Theorems

Non-uniform versions of the classical sampling theorem are applied to the compression of discrete two-dimensional (2D) signals. It is shown that the data rate of a hand-limited 2D signal can be reduced directly by sub-sampling methods. This means that the algorithm for compression is extremely simple, though the recovery of the original signal becomes complex. The procedure for reconstruction is derived by using linear filtering operations and 2D sampling rate alterations. The whole sub-sampling/reconstruction system is formulated in the form of a 2D multirate filter bank. Not every sub-sampling method allows for the perfect reconstruction of the original signal. A sufficient condition on the sub-sampling scheme is derived which would make reconstruction feasible. In theory, the reconstruction of the missing samples requires the use of piece-wise constant filters, but in practice their ideal responses can be approximated by FIR filters.

Generalized approach to sampling rate alteration in wave digital filters

Generalized arrangements (wave-flow diagrams) for sampling rate alteration in wave digital filters (WDFs) are proposed and analyzed. By this new approach, the number of degrees of freedom of the resulting so-called multirate WDF is not reduced in relation to that of the reference circuit. Advantages of the earlier multirate WDF's, such as improved loss performake for a given filter structure, increased dynamic range, and low passband sensitivity for coefficients, are preserved. Necessary and sufficient realiiability conditions for the generalized multirate WDF's are also presented. It is assumed that the reference filter is lossless, reciprocal, and rational in its complex frequency 4. An example of a fifth-order WDF for sampling rate increase is given.

Recursive Nth-band digital filters- Part I: Design and properties

We investigate the properties of a special class of recursive polyphase filters whose subfilters are distinct allpass filters. On the basis of the discovered properties, a Remez-type algorithm is constructed for designing these filters. The proposed algorithm is faster and gives higher filter selectivities than other existing methods. Better understanding of the filter properties allows us to select the branch filter orders such that the filter complexity is minimized. In addition to nonlinear-phase filters, the algorithm is applicable to the design of approximately linear-phase filters. Several examples are included illustrating the efficiency of the proposed design scheme. The characteristics of the resulting filters are exposed by means of experimental results, and the implementation of the allpass branch filters is discussed. In a companion paper [33], we use these polyphase filters as basic building blocks in constructing efficient filters for sampling rate alteration. In addition, it is shown how the don't care bands, which cannot be avoided in filters of this type, can be suppressed by using an additional correction filter stage.

Elliptic filter design for a class of generalized halfband filters

In multirate processing, symmetric finite impulse response halfband filters constitute an important class of filters that are useful for sampling rate alteration by a factor of two. Generalizing the notion of halfband filters to include infinite impulse response filters, a class of generalized halfband filters is examined and its properties are studied. The filters are efficient and have low roundoff noise. A filter design program has been developed to design a special class of elliptic filters taking into account restrictions due to the generalized halfband characteristics. The procedure adapts the results of elliptic filter design [10] to this special case. It is shown that if the passband and stopband edge frequencies are related, then these filters are computationally efficient even when no sampling rate change is needed.