A fast computation algorithm for the decision feedback equalizer

A novel fast algorithm for computing the minimum MSE decision feedback equalizer settings is proposed. The equalizer filters are computed indirectly, first by estimating the channel, and then by computing the coefficients in the frequency domain with the discrete Fourier transform (DFT). Approximating the correlation matrices by circulant matrices facilitates the whole computation with very small performance loss. The fractionally spaced equalizer settings are derived. The performance of the fast algorithm is evaluated through simulation. The effects of the channel estimation error and finite precision arithmetic are briefly analyzed. Results of simulation show the superiority of the proposed scheme     

Performance evaluation of a fast computation algorithm for the DMTin high-speed subscriber loop

The discrete multitone (DMT) modulation is considered to be a viable transmission scheme for high-speed subscriber loop. In this paper, the fast algorithm for computing the equalizer settings derived in [1] is extended and applied for the DMT in high-speed subscriber loop. The channel pulse response is assumed to be given by the channel identification method, and then the equalizer filter settings are computed. In simulations, a fast algorithm for the symbol spaced equalizer in a colored noise channel is used. Simulation results performed in various CSA loops indicate that the fast algorithm yields the near-optimum settings for the DMT system     

Perfect equalization for DMT systems without guard interval

We propose a new, low-complexity frequency-domain equalizer for discrete multitone (DMT) systems, which, in the absence of a guard interval, utilizes existing redundancy in the frequency-domain to completely eliminate intersymbol and interchannel interference. A perfect reconstruction condition is derived for the noise-free case leading to a sparse equalizer matrix structure. It is furthermore shown that under realistic scenarios minimum mean square error adaptation of the equalizer coefficients allows for nearly perfect reconstruction already for a much smaller amount of redundancy than indicated by the perfect reconstruction condition. The new equalization scheme has at least the same potential compared with traditional DMT while offering new degrees of freedom for designing short-latency DMT systems     

An efficient algorithm to compute the complete set of discreteGabor coefficients

The discrete Gabor (1946) transform algorithm is introduced that provides an efficient method of calculating the complete set of discrete Gabor coefficients of a finite-duration discrete signal from finite summations and to reconstruct the original signal exactly from the computed expansion coefficients. The similarity of the formulas between the discrete Gabor transform and the discrete Fourier transform enables one to employ the FFT algorithms in the computation. The discrete 1-D Gabor transform algorithm can be extended to 2-D as well.     

An efficient Fourier transform algorithm for multitone harmonicbalance

A novel Fourier transform technique is proposed for use in multitone harmonic-balance (HB) simulations. It is shown that computations of multitone distorted spectra reduce to efficient one-dimensional fast Fourier transform operations when certain relationships exist between the sampling rate and frequency components of the signal. The algorithm requires minimal initialization time and is readily incorporated into existing HE tools. It is especially useful when the number of input tones is very large, such as spectral regrowth and noise-power ratio simulations. The method is demonstrated on the example of a 5-GHz MESFET amplifier driven by a quadrature phase shift-keying modulated carrier     

分数阶傅立叶变换OFDM系统自适应均衡算法

由于子载波间干扰(ICI)的影响,传统OFDM系统均衡方法在快速衰落的信道环境下性能有较大下降.本文提出了一种基于分数阶傅立叶变换的OFDM系统自适应均衡方法,它用分数阶傅立叶变换代替傅立叶变换进行子载波调制与解调,同时在分数阶傅立叶域对接收信号进行自适应均衡.文中给出了最优分数阶傅立叶变换阶次的选取方法,和分数阶傅立叶域最小均方算法的步骤.分析和数值仿真结果表明,最优分数阶傅立叶域的自适应均衡算法较传统频域方法有更好的均衡效果,并且复杂度不高.     

一种基于分数阶傅里叶变换的OFDM系统及其均衡算法

在快速时变信道环境下,由于子载波间干扰(ICI)的影响,传统OFDM系统性能有较大下降.本文提出了一种基于分数阶傅里叶变换的OFDM系统,它用分数阶傅里叶变换代替傅里叶变换进行子载波调制与解调;同时,文中给出了最优分数阶傅里叶变换阶次的选取方法,并根据最小均方误差(MMSE)准则设计了分数阶傅里叶域乘性滤波器在接收端进行均衡.分析和数值仿真结果表明,最优分数阶傅里叶域的乘性滤波算法较频域方法有更好的均衡效果.     

Combined RLS-LMS initialization for per tone equalizers in DMT-receivers

In discrete multitone receivers, the classical equalizer structure consists of a (real) time domain equalizer (TEQ) combined with complex one-tap frequency domain equalizers. An alternative receiver is based on a per tone equalization (PTEQ), which optimizes the signal-to-noise ratio (SNR) on each tone separately and, hence, the total bitrate. In this paper, a new initialization scheme for the PTEQ is introduced, based on a combination of least mean squares (LMS) and recursive least squares (RLS) adaptive filtering. It is shown that the proposed method has only slightly slower convergence than full square-root RLS (SR-RLS) while complexity as well as memory cost are reduced considerably. Hence, in terms of complexity and convergence speed, the proposed algorithm is in between LMS and RLS.     

Bitrate-maximizing time-domain equalizer design for DMT-based systems

A time-domain equalizer (TEQ) is inserted in discrete multitone (DMT) receivers to impose channel shortening. Many algorithms have been developed to initialize this TEQ, but none of them really optimizes the bitrate. We present a truly bitrate-maximizing TEQ (BM-TEQ) cost function that is based on an exact formulation of the subchannel signal-to-noise ratio as a function of the TEQ taps. The performance of this BM-TEQ comes close to the performance of the per-tone equalizer.     

On implementing the arithmetic Fourier transform

The arithmetic Fourier transform (AFT), a method for computing the Fourier coefficients of a complex-valued periodic function, is based on a formula which has the advantage of eliminating many of the multiplications usually associated with computing discrete Fourier coefficients, but has the disadvantage of requiring samples of the signal at nonuniformly spaced time values. A method for computing the Fourier coefficients which allows uniform sampling at arbitrarily chosen sampling rates is developed. The technique still requires few multiplications, albeit at the expense of a limited amount of linear interpolation of the sample values. Efficient hardware implementations of this algorithm are presented     

Orthogonalization of circular stationary vector sequences and itsapplication to the Gabor decomposition

Certain vector sequences in Hermitian or in Hilbert spaces, can be orthogonalized by a Fourier transform. In the finite-dimensional case, the discrete Fourier transform (DFT) accomplishes the orthogonalization. The property of a vector sequence which allows the orthogonalization of the sequence by the DFT, called circular stationarity (CS), is discussed in this paper. Applying the DFT to a given CS vector sequence results in an orthogonal vector sequence, which has the same span as the original one. In order to obtain coefficients of the decomposition of a vector upon a particular nonorthogonal CS vector sequence, the decomposition is first found upon the equivalent DFT-orthogonalized one and then the required coefficients are found through the DFT. It is shown that the sequence of discrete Gabor (1946) basis functions with periodic kernel and with a certain inner product on the space of N-periodic discrete functions, satisfies the CS condition. The theory of decomposition upon CS vector sequences is then applied to the Gabor basis functions to produce a fast algorithm for calculation of the Gabor coefficients     

Discrete Gabor transform

A feasible algorithm for implementing the Gabor expansion, the coefficients of which are computed by the discrete Gabor transform (DGT), is presented. For a given synthesis window and sampling pattern, computing the auxiliary biorthogonal function of the DGT is nothing more than solving a linear system. The DGT presented applies for both finite as well as infinite sequences. By exploiting the nonuniqueness of the auxiliary biorthogonal function at oversampling an orthogonal like DGT is obtained. As the discrete Fourier transform (DFT) is a discrete realization of the continuous-time Fourier transform, similarly, the DGT introduced provides a feasible vehicle to implement the useful Gabor expansion     

Computation of an odd-length DCT from a real-valued DFT of the samelength

The discrete cosine transform (DCT) is often computed from a discrete Fourier transform (DFT) of twice or four times the DCT length. DCT algorithms based on identical-length DFT algorithms generally require additional arithmetic operations to shift the phase of the DCT coefficients. It is shown that a DCT of odd length can be computed by an identical-length DFT algorithm, by simply permuting the input and output sequences. Using this relation, odd-length DCT modules for a prime factor DCT are derived from corresponding DFT modules. The multiplicative complexity of the DCT is then derived in terms of DFT complexities     

An analog speech scrambling system using the FFT technique with high-level security

A full-duplex analog speech-scrambling system is proposed for application to mobile communication systems and public switched telephone networks. The scrambling algorithm is based on the rearrangement of the fast Fourier transform (FFT) coefficients accompanied by adaptive dummy spectrum insertion and companding operation. The simulation results indicate that the scrambled speech has no residual intelligibility and the descrambled speech quality is satisfactory. The hardware unit is implemented by using seven advanced digital signal processor chips, including those for an adaptive equalizer and an echo canceller for full-duplex operation. The performance is proven to be satisfactory.<>     

Perfect discrete multitone modulation with optimal transceivers

Previously, discrete Fourier transform (DFT)-based discrete multitone modulation (DMT) systems have been widely applied to various applications. In this paper, we study a broader class of DMT systems using more general unitary matrices instead of DFT matrices. For this class, we show how to design optimal DMT systems over frequency-selective channels with colored noise. In addition, asymptotical performance of DFT-based and optimal DMT systems are studied and shown to be equivalent. However, for a moderate number of bands, the optimal DMT system offers significant gain over the DFT-based DMT system, as is demonstrated by examples     

Optimal impulse response shortening for discrete multitonetransceivers

The authors have derived a new algorithm for the optimal shortening of a channel impulse response in discrete multitone (DMT) transceivers. This algorithm uses eigenvalues and eigenvectors to generate the coefficients of the shortening impulse response filter (SIRF). In comparison with the previous approach, this new algorithm can calculate the optimal settings of an SIRF with arbitrary length     

A new texture generation method based on pseudo-DCT coefficients.

In this paper, a new method for generating different texture images is presented. This method involves a simple transform from a certain one-dimensional (1-D) signal to an expected two-dimensional (2-D) image. Unlike traditional methods, the input signal is generated by a simple 1-D function in our work instead of a sample texture. We first transform the 1-D input signal into frequency domain using fast Fourier transform. Based on the sufficient analysis in 2-D discrete cosine transform (DCT) domain, where each of the coefficients expresses a texture feature in a certain direction, the 2-D pseudo-DCT coefficients are then constructed by appropriately rearranging the Fourier coefficients in terms of their frequency components. Finally, the corresponding texture image can be produced by 2-D inverse DCT algorithm. We applied the proposed method to generate several stochastic textures (i.e., cloud, illumination, and sand), and several structural texture images. Experimental results indicate the good performance of the proposed method.     

Joint window and time domain equalizer design for bit rate maximization in DMT-receivers

In discrete multitone (DMT) receivers, as for instance in asymmetric digital subscriber lines (ADSLs), the classical equalizer structure consists of a (real) time domain equalizer (TEQ) combined with complex 1-tap frequency domain equalizers (FEQs). Additionally, receiver windowing can be applied to mitigate the bad spectral containment of the demodulating DFT sidelobes. We focus on a combined equalizer and windowing design procedure to maximize the achievable bit rate in DMT-based modems. Whereas the combination of a TEQ with a single window treats all the data carrying tones in a common way, the presented design method can also be used in a "per group" fashion, where smaller groups of tones receive each a different equalizer-window pair. When such groups contain only one single tone, the design procedure can be linked to the performance of an unbiased minimum mean square error (MMSE) per tone equalizer (PTEQ), which then also implicitly implements a per tone window. The general framework introduced Allows us to treat equalizer-only and window-only designs as well, which appear as special cases in a natural way. This set of bit rate maximizing techniques can serve either as practical design methods or as upper bounds for existing (suboptimal) methods. We will also show that for the same achievable bit rate, equalizer taps can be exchanged for windowing coefficients to reduce complexity during data transmission.     

Adaptive bit rate maximizing time-domain equalizer design for DMT-based systems

The classical discrete multitone receiver as used in, e.g., digital subscriber line (DSL) modems, combines a channel shortening time-domain equalizer (TEQ) with one-tap frequency-domain equalizers (FEQs). In a previous paper, the authors proposed a nonlinear bit rate maximizing (BM) TEQ design criterion and they have shown that the resulting BM-TEQ and the closely related BM per-group equalizers (PGEQs) approach the performance of the so-called per-tone equalizer (PTEQ). The PTEQ is an attractive alternative that provides a separate complex-valued equalizer for each active tone. In this paper, the authors show that the BM-TEQ and BM-PGEQ, despite their nonlinear cost criterion, can be designed adaptively, based on a recursive Levenberg-Marquardt algorithm. This adaptive BM-TEQ/BM-PGEQ makes use of the same second-order statistics as the earlier presented recursive least-squares (RLS)-based adaptive PTEQ. A complete range of adaptive BM equalizers then opens up: the RLS-based adaptive PTEQ design is computationally efficient but involves a large number of equalizer taps; the adaptive BM-TEQ has a minimal number of equalizer taps at the expense of a larger design complexity; the adaptive BM-PGEQ has a similar design complexity as the BM-TEQ and an intermediate number of equalizer taps between the BM-TEQ and the PTEQ. These adaptive equalizers allow us to track variations of transmission channel and noise, which are typical of a DSL environment.     

Algorithmes de transformations de Fourier et en cosinus mono et bidimensionnels

This paper presents a fast algorithm for the computation of the discrete Fourier and cosine transform, and this for transform lengths which are powers of 2. This approach achieves the lowest known number of operations (multiplications and additions) for the discrete Fourier transform of real, complex, symmetrical and antisymmetrical sequences, for the odd discrete Fourier transform and for the discrete cosine transform. The extension to the two-dimensional Fourier and cosine transform is presented as well.