Recent years have seen the development of signal denoising algorithms based on wavelet transform. It has been shown that thresholding the wavelet coefficients of a noisy signal allows to restore the smoothness of the ...
详细信息
ISBN:
(纸本)0780370414
Recent years have seen the development of signal denoising algorithms based on wavelet transform. It has been shown that thresholding the wavelet coefficients of a noisy signal allows to restore the smoothness of the original signal. However, wavelet denoising suffers of a main drawback : around discontinuities the reconstructed signal is smoothed, exhibiting pseudo-Gibbs phenomenon. We consider the problem of denoising piecewise smooth signals with sharp discontinuities. We propose to apply a traditional wavelet denoising method and to restore the denoised signal using a total variation minimization approach. This second step allows to remove the Gibbs phenomena and therefore to restore sharp discontinuities, while the other structures are preserved. The main innovation of our algorithm is to constrain the total variation minimization by the knowledge of the remaining wavelet coefficients. In this way, we make sure that the restoration process does not deteriorate the information that has been considered as significant in the denoising step. With this approach we substantially improve the performance of classical wavelet denoising algorithms, both in terms of SNR and in terms of visual artifacts.
Real-time SAR processing requires high computational power. At the Laboratorium fur Informationstechnologie a parallel DSP called HiPAR-DSP 16 was developed, which is optimized for imageprocessingalgorithms. In this...
详细信息
ISBN:
(纸本)0819440612
Real-time SAR processing requires high computational power. At the Laboratorium fur Informationstechnologie a parallel DSP called HiPAR-DSP 16 was developed, which is optimized for imageprocessingalgorithms. In this paper we present a compact multi-DSP board utilizing the HiPAR-DSP 16. The board can deliver a performance of up to 15 GOPS, has a volume of 160x230x20 mm, and consumes less than 20 W. The first version of the board is used for filtering of SAR data to show the capabilities of the system. An estimation showed that our board can process an omegak algorithm with a rangeline length of 4096 samples and a PRF of 600 Hz in real-time.
It would be desirable, in terms of energy conservation, to use a low complexity approximate algorithm to do all DCT and IDCT computation in an MPEG-2 video codec. However, there is a significant quality penalty associ...
详细信息
ISBN:
(纸本)0780370414
It would be desirable, in terms of energy conservation, to use a low complexity approximate algorithm to do all DCT and IDCT computation in an MPEG-2 video codec. However, there is a significant quality penalty associated with this approach that may not always be acceptable. A practical algorithmic method is studied here for achieving scalable energy reduction during DCT and IDCT computation in MPEG-2 video codecs at the expense of reasonable amounts of quality. For example, by applying exact and approximate DCT/IDCT algorithms appropriately, the energy consumption of DCT and IDCT execution in two video codecs communicating with one another can be reduced by 8% for quality reduction of 0.4 dB average PSNR, 14% for 0.8 dB reduction, or 22% for 1.4 dB reduction.
CDMA systems in multipath fading channels need to estimate channel parameters for coherent detection of the transmitted signals. In this paper we present a simple but effective channel estimation algorithm that can be...
详细信息
ISBN:
(纸本)0780370414
CDMA systems in multipath fading channels need to estimate channel parameters for coherent detection of the transmitted signals. In this paper we present a simple but effective channel estimation algorithm that can be incorporated into most types of multiuser receivers to obtain good detection performance. This technique uses a set of correlation filters to independently estimate each of the channel parameters. One advantage our method has over subspace-based algorithms for channel estimation is that it can estimate the channel parameters without phase or amplitude ambiguity. Simulation results demonstrating that our channel estimator is capable of tracking reasonably fast fading channels are also presented in the paper.
We consider the problem of estimating continuous-time autoregressive (CAR) processes from discrete-time noisy observations. This can be done within a Bayesian framework using Markov chain Monte Carlo (MCMC) methods. E...
详细信息
ISBN:
(纸本)0780370414
We consider the problem of estimating continuous-time autoregressive (CAR) processes from discrete-time noisy observations. This can be done within a Bayesian framework using Markov chain Monte Carlo (MCMC) methods. Existing methods include the standard random walk Metropolis algorithm. On the other hand, least-squares (LS) algorithms exist where derivatives are approximated by differences and parameter estimation is done in a least-squares manner. In this paper, we incorporate the LS estimation into the MCMC framework to develop a new MCMC algorithm. This new algorithm is combined with the standard Metropolis algorithm and is found to improve performance compared to the standard MCMC algorithm. Simulation results are presented to support our findings.
In this paper we present an optimized DSP implementation of a modified error-feedback lattice least-square (EF-LSL) adaptive filtering algorithm. Simple measures that provide numerical stability for poor persistent ex...
详细信息
ISBN:
(纸本)0780370414
In this paper we present an optimized DSP implementation of a modified error-feedback lattice least-square (EF-LSL) adaptive filtering algorithm. Simple measures that provide numerical stability for poor persistent excitation are also proposed. As a result of the optimization and the stability measures, an efficient and stable implementation of a fast algorithm of the RLS family was attained. We present the results of an acoustic echo cancelling experiment performed with the implemented algorithm. With a 40 MIPS SHARC DSP, up to 290 adaptive filter coefficients can be used. This represents an effective alternative to algorithms of the LMS family, while still retaining the good convergence properties of the RLS family.
In digital mobile communications efficient compression algorithms are needed to encode speech or audio signals. As the determined source parameters are highly sensitive to transmission errors, robust source and channe...
详细信息
ISBN:
(纸本)0780370414
In digital mobile communications efficient compression algorithms are needed to encode speech or audio signals. As the determined source parameters are highly sensitive to transmission errors, robust source and channel decoding schemes are required. This contribution deals with an iterative source-channel decoding approach where a simple channel decoder and a softbit-source decoder are concatenated. We will mainly focus on softbit-source decoding which can be considered as error concealment technique. This technique utilizes residual redundancy remaining after source coding. In this paper we derive a new formula that shows how the residual redundancy transforms into the extrinsic information utilizable for iterative decoding. The derived formula opens several starting points for optimizations, e.g. it helps to find a robust index assignment. Furthermore, it allows the conclusion that softbit-source decoding is the limiting factor if applied to iterative decoding processes. Therefore, no significant gain will be obtainable by more than two iterations. This will be demonstrated by simulation.
In this paper, we present a high-level synthesis technique targeting low power consumption for data-dominated applications. We have used a statistical estimation technique to obtain switching activity of modules when ...
详细信息
ISBN:
(纸本)0780370414
In this paper, we present a high-level synthesis technique targeting low power consumption for data-dominated applications. We have used a statistical estimation technique to obtain switching activity of modules when sharing of computing resources are required in a design. The technique enables us to understand switching behavior under resource sharing. Using the relationship between switching power and resource sharing thus obtained, we developed scheduling and allocation algorithms to reduce data path switching power. Experiments performed on various examples show up to 49% improvement in power reduction under resource constraints.
Next generation reconnaissance and Automatic Target Detection/Recognition (ATD/R) performance goals will impose new image quality requirements on integrated SAR hardware and software systems. Signal processing techniq...
详细信息
ISBN:
(纸本)0819440698
Next generation reconnaissance and Automatic Target Detection/Recognition (ATD/R) performance goals will impose new image quality requirements on integrated SAR hardware and software systems. Signal processing techniques using demonstrated non-parametric autofocus methods such as the Phase Gradient Autofocus (PGA) algorithm and developments in robust super-resolution signal processing offer the opportunity for reducing overall system cost through utilization of less costly hardware options in integrated system design. Traditional requirements on image quality from integrated hardware-software SAR systems have used image quality metrics based on the characteristics of the overall system impulse response function. An additional class of image quality metrics is available based on the performance of the ATD/R algorithms that are to utilize the imagery. The performance of a given SAR system by these measures is expected to be context-sensitive and dependant on both target and clutter characteristics in a manner not necessarily readily characterizable solely in terms of system impulse response function measures of image quality. A simulation illustration of these issues is presented for a test case in which a range of SAR sensor hardware options are processed through a representative texture metric mechanization. Potential performance dependencies on target and clutter characteristics are reviewed and the efficacy of supplementing impulse response function image quality metrics with additional appropriate predictors of ATD/R performance is reviewed.
This paper describes two approaches suitable for an FPGA implementation of Walsh-Hadamard transforms. These transforms are important in many signal-processing applications including speech compression, filtering and c...
详细信息
ISBN:
(纸本)0780370414
This paper describes two approaches suitable for an FPGA implementation of Walsh-Hadamard transforms. These transforms are important in many signal-processing applications including speech compression, filtering and coding. Two novel architectures for the Fast Hadamard Transforms using both systolic architecture and distributed arithmetic techniques are presented. The first approach uses the Baugh-Wooley multiplication algorithm for a systolic architecture implementation. The second approach is based on both distributed arithmetic ROM and accumulator structure, and a sparse matrix factorisation technique. Implementations of the algorithms on a Xilinx FPGA board are described. Distributed arithmetic approach exhibits better performances when compared with the systolic architecture approach.
暂无评论