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1996 8th European Signal Processing Conference (EUSIPCO 1996)最新文献

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A new robust adaptive step size LMS algorithm 一种新的鲁棒自适应步长LMS算法
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.35932
D. I. Pazaitis, A. Constantinides
In this contribution a new robust technique for adjusting the step size of the Least Mean Squares (LMS) adaptive algorithm is introduced. The proposed method exhibits faster convergence, enhanced tracking ability and lower steady state excess error compared to the fixed step size LMS and other previously developed variable step size algorithms, while retaining much of the LMS computational simplicity. A theoretical behaviour analysis is conducted and equations regarding the evolution of the weight error vector correlation matrix together with convergence bounds are established. Extensive simulation results support the theoretical analysis and confirm the desirable characteristics of the proposed algorithm.
本文介绍了一种新的鲁棒技术,用于调整最小均方(LMS)自适应算法的步长。与固定步长LMS和其他已开发的变步长LMS算法相比,该方法具有更快的收敛速度、更强的跟踪能力和更小的稳态过量误差,同时保留了LMS计算的简便性。进行了理论行为分析,建立了权值误差矢量相关矩阵的演化方程和收敛界。大量的仿真结果支持了理论分析,并证实了所提出算法的理想特性。
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引用次数: 2
Low bit rate video coding for mobile multimedia communications 用于移动多媒体通信的低比特率视频编码
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36393
R. Lagendijk, J. Biemond, Cor P. Quist
In this paper we first describe the objectives of the Delft Mobile Multimedia Communications project. Next, the subject of lossy contour compression is considered in more detail as it is an essential component of most object or region-based compression techniques for low bit rate video coding. We propose an optimized B-splines approximation approach, which results in a 40 percent higher compression than the lossless conditional chain code method. Achieved rates are, depending on the tolerable deviation between original and coded contour, in the order of 0.70 to 0.90 bit per contour pixel.
在本文中,我们首先描述了代尔夫特移动多媒体通信项目的目标。接下来,将更详细地考虑有损轮廓压缩的主题,因为它是大多数低比特率视频编码的基于对象或区域的压缩技术的重要组成部分。我们提出了一种优化的b样条近似方法,其压缩比无损条件链码方法高40%。根据原始轮廓和编码轮廓之间的可容忍偏差,实现的速率为每轮廓像素0.70至0.90比特。
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引用次数: 10
Statistical LIP modelling for visual speech recognition 视觉语音识别的统计LIP建模
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36365
J. Luettin, N. Thacker, S. W. Beet
We describe a speechreading (lipreading) system purely based on visual features extracted from grey level image sequences of the speaker's lips. Active shape models are used to track the lip contours while visual speech information is extracted from the shape of the contours. The distribution and temporal dependencies of the shape features are modelled by continuous density Hidden Markov Models. Experiments are reported for speaker independent recognition tests of isolated digits. The analysis of individual feature components suggests that speech relevant information is embedded in a low dimensional space and fairly robust to inter- and intra-speaker variability.
我们描述了一个纯粹基于从说话者嘴唇灰度图像序列中提取的视觉特征的语音读(唇读)系统。主动形状模型用于跟踪唇轮廓,同时从轮廓形状中提取视觉语音信息。利用连续密度隐马尔可夫模型对形状特征的分布和时间依赖性进行建模。本文报道了独立于说话人的孤立数字识别实验。对单个特征分量的分析表明,语音相关信息嵌入在低维空间中,对说话人之间和说话人内部的变化具有相当强的鲁棒性。
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引用次数: 31
Multiscale edges detection algorithm implementation using FPGA devices 用FPGA器件实现多尺度边缘检测算法
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36058
M. Paindavoine, Sarifuddin, C. Milan, J. Grapin
One of the way to extract edges uses the fast wavelet transform algorithm. This technique allows the detection of multiscale edge and is used to detect all the details which are in a picture by modifying the scale. The real time application for edge detection involves the implementation of the algorithm on an integrated circuit like an FPGA and the development of an appropriated board. This article deals about the implementation of a wavelet transform algorithm onto a FPGA and the development of an electronic board to detect multiscale edges.
其中一种提取边缘的方法是快速小波变换算法。该技术允许检测多尺度边缘,并通过修改尺度来检测图像中的所有细节。边缘检测的实时应用涉及在FPGA等集成电路上实现算法和开发适当的板。本文讨论了小波变换算法在FPGA上的实现和多尺度边缘检测电路板的开发。
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引用次数: 1
Perceptual coding of speech using a fast wavelet packet transform algorithm 语音感知编码采用快速小波包变换算法
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36342
B. Carnero, A. Drygajlo
This paper presents a new speech coding algorithm based on a fast wavelet packet transform algorithm and psychoacoustic modeling. The employed FFT-like overlapped block orthogonal transform allows us to approximate the auditory critical band decomposition in an efficient manner, which is a major advantage over previous approaches. Owing to such a decomposition of the original signal, we make use of the human ear masking properties to decrease the mean bit rate of the encoder.
提出了一种基于快速小波包变换算法和心理声学建模的语音编码算法。所采用的类似fft的重叠块正交变换使我们能够以一种有效的方式近似听觉临界带分解,这是比以往方法的主要优点。由于原始信号的这种分解,我们利用人耳掩蔽特性来降低编码器的平均比特率。
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引用次数: 3
A rational filter for the removal of blocking artifacts in image sequences coded at low bitrate 一种合理的滤波器,用于去除低比特率编码的图像序列中的阻塞伪影
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36037
R. Castagno, G. Ramponi
In this paper a simple but effective operator for the reduction of blocking artifacts is presented. The method is based on the Rational Filter approach: the operator is expressed as a ratio between a linear and a polynomial function of the input data. Such filters proved to outperform other conventional methods in other applications, such as noise smoothing [1], thanks to their capability of adapting gradually to the local image characteristics. The filter is capable of biasing its behaviour in order to achieve good performance both in uniform areas, where linear smoothing is needed, and in textured zones, where nonlinear and directional filtering is required. A detector of activity is embedded in the expression of the operator itself so that the biasing of the behaviour of the filter is smooth and not based on fixed thresholds. The proposed method has been originally designed as a post-processing tool for frames of sequences coded at medium-low bitrate with the dynamic coding scheme proposed in [2], but gave good results also when applied to JPEG coded images.
本文提出了一种简单而有效的减少块伪影的算子。该方法基于Rational Filter方法:操作符表示为输入数据的线性函数和多项式函数之间的比率。由于这种滤波器能够逐渐适应局部图像特征,因此在其他应用中,如噪声平滑[1],证明其性能优于其他传统方法。该滤波器能够对其行为进行偏置,以便在需要线性平滑的均匀区域和需要非线性和定向滤波的纹理区域实现良好的性能。活动检测器嵌入在算子本身的表达式中,因此滤波器的行为偏差是平滑的,而不是基于固定的阈值。该方法最初被设计为使用[2]中提出的动态编码方案以中低比特率编码的序列帧的后处理工具,但在应用于JPEG编码图像时也取得了良好的效果。
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引用次数: 21
Coherent model-based optical resolution and SNR 基于相干模型的光学分辨率和信噪比
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36325
A. J. D. Dekker
In this paper a new parameter estimation based criterion for two-point resolution is proposed. Unlike the classical resolution criteria, the new criterion takes account of noise and systematic errors. A resolution limit in terms of the observations is derived. This limit depends on the point spread function used and the degree of coherence supposed. For statistical observations the probability of resolution as a function of the SNR is derived. This probability can be used as a performance measure in the assessment of optical instruments.
本文提出了一种新的基于参数估计的两点分辨准则。与传统的分辨率判据不同,新判据考虑了噪声和系统误差。根据观测结果导出了分辨率极限。这个极限取决于所使用的点扩展函数和假定的相干度。对于统计观测,导出了作为信噪比函数的分辨率概率。这一概率可以作为光学仪器评估的性能指标。
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引用次数: 0
Parallel implementation of image coding usingwavelet transform: Syndex software environment application 用小波变换并行实现图像编码:Syndex软件环境应用
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36261
C. Cudel, B. Vigouroux
This work is a contribution to Adequation between Algorithm and Architecture. It presents an example of application made with SynDEx, a software environment to implement signal processing or automatic algorithms on multiprocessor network. This communication shows that a Conditioned Data Flow Graph used for modelling an algorithm, is enough to do an implementation on multi-processeur network.
这项工作是对算法和架构之间的不足的贡献。给出了一个应用SynDEx软件环境在多处理器网络上实现信号处理或自动算法的实例。这种通信表明,用于建模算法的条件数据流图足以在多处理器网络上实现。
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引用次数: 0
Steady-state performance analysis of the LMS adaptive time-varying second order Volterra filter LMS自适应时变二阶Volterra滤波器的稳态性能分析
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36117
M. Sayadi, F. Fnaiech, S. Guillon, M. Najim
In this paper, the steady-state performance of the Least Mean Square (LMS) adaptive second order Volterra filter, with constant step-sizeμ, in a time-varying setting, is analysed. The quantitative evaluation of the steady-state Excess Mean Square Error (EMSE), where the contribution of the gradient misadjustment and the tracking error are well characterized, is established. The optimum step-size for time-varying second order Volterra filter is then given. Thus, we can study the correlation between the Excess MSE and the optimum step-size in one hand and the parameters of the time-varying nonlinear system, in the other hand. Furthermore, the steady-state behavior predicted by the analysis is in good agreement with the experimental results. The adaptive filter was used in a second order Volterra system identification in a non stationary environment.
本文分析了定步长μ的最小均方自适应二阶Volterra滤波器在时变条件下的稳态性能。建立了稳态超均方误差(EMSE)的定量评价,其中梯度失调和跟踪误差的贡献得到了很好的表征。给出了时变二阶Volterra滤波器的最优步长。因此,我们一方面可以研究过剩均方差与最优步长之间的关系,另一方面也可以研究时变非线性系统参数之间的关系。此外,分析预测的稳态行为与实验结果吻合较好。将自适应滤波器应用于非平稳环境下的二阶Volterra系统辨识。
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引用次数: 2
RBF networks for density estimation 密度估计的RBF网络
Pub Date : 1996-09-01 DOI: 10.5281/ZENODO.36129
L. Sardo, J. Kittler
A non-parametric probability density function (pdf) estimation technique is presented. The estimation consists in approximating the unknown pdf by a network of Gaussian Radial Basis Functions (GRBFs). Complexity analysis is introduced in order to select the optimal number of GRBFs. Results obtained on real data show the potentiality of this technique.
提出了一种非参数概率密度函数估计技术。该估计包括用高斯径向基函数网络(GRBFs)逼近未知的pdf。为了选择最优的grbf数量,引入了复杂性分析。实际数据的结果表明了该技术的潜力。
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引用次数: 1
期刊
1996 8th European Signal Processing Conference (EUSIPCO 1996)
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