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

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Detection of abrupt changes: A time-frequency approach 突变检测:一种时频方法
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.35977
H. Laurent, C. Doncarli, P. Poignet
This paper presents a comparison between parametric and non-parametric approaches of abrupt changes detection in noisy signals. The goal is to propose an alternative way to be used when the model-based methods do not work very well because of an unsuitable model structure or a non strictly stationnary stepwise signal. In this latter case, an analysis of time-frequency distributions allows the detection of abrupt spectral changes without any hypothesis and provides some results as good as parametric methods for the studied type of signals.
本文对噪声信号突变检测的参数方法和非参数方法进行了比较。目标是当基于模型的方法由于模型结构不合适或非严格平稳逐步信号而不能很好地工作时,提出一种替代方法。在后一种情况下,对时频分布的分析可以在没有任何假设的情况下检测到突然的频谱变化,并且对所研究的信号类型提供一些与参数方法一样好的结果。
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引用次数: 0
Multi-channel adaptive filtering applied to multi-channel acoustic echo cancellation 多通道自适应滤波应用于多通道回声消除
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36426
J. Benesty, P. Duhamel, Y. Grenier
This paper presents some new ways of deriving multichannel (M-C) adaptive algorithms in the context of M-C acoustic echo cancellation (AEC).
本文提出了在多通道回声抵消(AEC)背景下推导多通道(M-C)自适应算法的新方法。
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引用次数: 11
A multivariable Steiglitz-McBride method 一个多变量Steiglitz-McBride方法
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36063
M. Ashari, M. Mboup, P. Regalia
In this paper, we present an off-line multi-input/multi-output version of the Steiglitz-McBride method, as well as an analytic description of the set of its stationary points. As in the scalar case [13], the description is given in terms of first- and second-order interpolation constraints, respectively, on the model impulse response and covariance sequences. The constraints are related to the theory of g-Markov covariance equivalent realizations and generalize the work of Inouye [7] and King et al. [9].
本文给出了Steiglitz-McBride方法的一种离线多输入/多输出版本,并给出了该方法的平稳点集的解析描述。与标量情况[13]一样,分别用一阶和二阶插值约束对模型脉冲响应和协方差序列进行描述。这些约束与g-Markov协方差等价实现理论有关,并推广了Inouye[7]和King等人[7]的工作。
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引用次数: 0
Cell delay variation performance of CBR and VBR MPEG-2 sources in an atm multiplexer atm多路复用器中CBR和VBR MPEG-2源的小区延迟变化性能
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36451
Javier Zamora, D. Anastassiou, Kand Ly
Video services require specific constraints regarding the delay variation or jitter experienced when they are transmitted in packet networks such as ATM. This delay component is mainly generated in multiplexing processes and it has a direct impact on the final QoS. In this paper the jitter issue is addressed in the environment of a video server connected to an ATM Network. Both CBR and VBR MPEG-2 streams are considered as traffic sources. For each video source its delay variation is studied using first order and second order statistics such as jitter variance and GCRA, respectively. We study several traffic scenarios, where correlation between video sources is considered. Finally the obtained results are compared with the M+D/D/l model.
视频业务在分组网络(如ATM)中传输时,需要对延迟变化或抖动进行特定的限制。该延迟分量主要产生于复用过程中,直接影响到最终的QoS。本文针对ATM网络视频服务器环境中的抖动问题进行了研究。CBR和VBR的MPEG-2流都被认为是流量源。对于每个视频源,分别使用一阶和二阶统计量(如抖动方差和GCRA)研究其延迟变化。我们研究了几种交通场景,其中考虑了视频源之间的相关性。最后将所得结果与M+D/D/l模型进行了比较。
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引用次数: 8
A wide-band speech-model process as a test signal 一个宽带语音模型过程作为测试信号
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36362
M. R. Serafat, U. Heute
Some of the major problems in objective quality assessment of speech coding systems or in testing other adaptive speech transmission systems are the speaker dependence, reproducibility, and the comparability of the measurement results, if natural speech is used as the test signal. This problem can be avoided by using suitable speech-model processes. In this paper, we present a wide-band speech-model process, which includes the same long- and short-time characteristics as natural speech. The controlling part of the generator of this process involves several trained Markov chains (mc) to adapt the time-varying properties of the process to those of natural speech. Furthermore, special care is taken of the necessary probabilty density function (PDF) asymmetries, because the natural wide-band speech has an asymmetric PDF.
如果使用自然语音作为测试信号,语音编码系统的客观质量评估或其他自适应语音传输系统测试中的一些主要问题是说话人的依赖性、再现性和测量结果的可比性。使用合适的语音模型处理可以避免这个问题。在本文中,我们提出了一种宽频带语音模型过程,它具有与自然语音相同的长、短时特征。该过程的生成器的控制部分包括几个训练好的马尔可夫链(mc),以使过程的时变特性适应于自然语音的时变特性。此外,由于自然宽带语音具有不对称的概率密度函数(PDF),因此特别注意必要的概率密度函数不对称性。
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引用次数: 2
A fuzzy expert system for low level image segmentation 低级图像分割的模糊专家系统
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36431
M. Barni, S. Rossi, A. Mecocci
In this paper a general purpose fuzzy expert system is presented for low level image segmentation. By means of approximate reasoning based on fuzzy logic, the criticality of the choice of the several thresholds and parameters which usually must be tuned to make the expert system work properly is reduced. More specifically, it is proved that, by keeping constant the number of rules the expert system consists of, the fuzzy approach permits to build a more general system, capable of giving satisfactory results for a large number of images stemming from different applications. The validity of the approach is demonstrated by comparing the effectiveness of a classical expert system with that of its corresponding fuzzy version. Upon analysis of the results, the superiority of the fuzzy system in terms of robustness and generality comes out.
本文提出了一种用于低级图像分割的通用模糊专家系统。通过基于模糊逻辑的近似推理,降低了专家系统正常工作所需的几个阈值和参数选择的临界性。更具体地说,证明了通过保持专家系统所包含的规则数量不变,模糊方法允许构建一个更通用的系统,能够为来自不同应用的大量图像提供满意的结果。通过将经典专家系统的有效性与相应的模糊专家系统的有效性进行比较,证明了该方法的有效性。通过对结果的分析,得出了模糊系统在鲁棒性和通用性方面的优越性。
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引用次数: 6
Discrete models for multidimensional system simulation 多维系统仿真的离散模型
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36068
R. Rabenstein
Multidimensional continuous systems arising from physical applications with distributed parameters are conventionally modelled by partial differential equations. This paper presents an alternate description by transfer functions based on suitably chosen functional transformations. Signal processing techniques lead to discrete simulation models which are suitable for computer implementation. Numerical results show considerable savings in computer time over existing numerical methods.
由具有分布参数的物理应用产生的多维连续系统通常由偏微分方程建模。本文提出了一种基于适当选择的函数变换的传递函数替代描述方法。信号处理技术产生了适合于计算机实现的离散仿真模型。数值结果表明,与现有的数值方法相比,计算机时间大大节省。
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引用次数: 6
Implementation of a European paging system receiver using CORDIC algorithm 使用CORDIC算法的欧洲寻呼系统接收器的实现
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36061
J. Vuori, J. Skyttä
ERMES is a new European paging standard. The data transmission speed is higher than in older systems, e. g. POCSAG, and advanced new features are implemented including intelligent battery saving operation and country roaming. The higher speed is achieved using the more elaborate modulation method 4-PAM/FM which makes the demodulator implementation much harder than in older 2-FSK based paging systems. The objective of this paper is to propose a novel ERMES signal demodulator structure utilising a complex digital phase-locked loop which is implemented using the CORDIC algorithm. Phase-locked loop demodulators have inherently better performance than the normally used discriminator type detectors. Implementation of those phase-locked loop structures using the CORDIC algorithm makes VLSI realizations very feasible.
ERMES是一个新的欧洲寻呼标准。数据传输速度高于POCSAG等旧系统,并实现了智能省电操作和国家漫游等先进新功能。使用更复杂的调制方法4-PAM/FM实现更高的速度,这使得解调实现比旧的基于2-FSK的寻呼系统困难得多。本文的目的是提出一种新的ERMES信号解调器结构,该结构利用CORDIC算法实现了一个复杂的数字锁相环。锁相环解调具有比通常使用的鉴别器型检测器更好的性能。使用CORDIC算法实现这些锁相环结构使得VLSI的实现非常可行。
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引用次数: 0
Locally adaptive techniques for stack filtering 堆栈滤波的局部自适应技术
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36042
D. Petrescu, I. Tabus, M. Gabbouj
This paper introduces a new structure for stack filtering, where the filter adapts to the local characteristics encountered in data. Both supervised and unsupervised techniques for optimal design are investigated. We split the image into small regions and select the stack filter to process each region according to the spatial domain or threshold level domain characteristics of the input signal. This method provides a significant improvement potential over the global stack filtering approach. Some local statistics are computed, to build a reduced input space which efficiently describes the most important local characteristics of data. Vector quantization is used for clustering the reduced input space into a small number of regions, and then finding a mapping between reduced input space clusters and the filter space, will result in rules for selecting the best suited stack filter for a given region. The supervised clustering procedures are shown to surpass significantly the global filtering approach.
本文介绍了一种新的堆栈滤波结构,该滤波器能够适应数据中遇到的局部特征。研究了有监督和无监督两种优化设计方法。我们将图像分割成小区域,并根据输入信号的空间域或阈值水平域特征选择叠加滤波器对每个区域进行处理。与全局堆栈过滤方法相比,该方法提供了显著的改进潜力。计算一些局部统计量,以建立一个简化的输入空间,有效地描述数据的最重要的局部特征。矢量量化用于将简化后的输入空间聚类为少量区域,然后找到简化后的输入空间聚类与滤波器空间之间的映射,从而得到为给定区域选择最适合的堆栈滤波器的规则。有监督聚类方法明显优于全局过滤方法。
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引用次数: 4
Nonlinear prediction of speech signals using radial basis function networks 基于径向基函数网络的语音信号非线性预测
Pub Date : 1996-09-10 DOI: 10.5281/ZENODO.36355
M. Birgmeier
In this paper, we compare the capabilities of various forms of radial basis function networks as nonlinear short-term predictors for speech signals representing sustained utterances of German vowels. We use RBF and RBF-AR1 network architectures, trained using a standard algorithm or alternatively the extended Kalman filter (EKF) algorithm, and linear least squares predictors. We also look at cascaded forms of linear/nonlinear predictors. We evaluate both prediction gain and spectral flatness measure of the residual. The results indicate: The RBF-AR structure is the most powerful, EKF training yields better results than standard training for RBF networks, and a non-cascaded RBF-AR predictor produces results superior to cascaded predictors.
在本文中,我们比较了各种形式的径向基函数网络作为代表德语元音持续话语的语音信号的非线性短期预测器的能力。我们使用RBF和RBF- ar1网络架构,使用标准算法或扩展卡尔曼滤波(EKF)算法和线性最小二乘预测器进行训练。我们还研究了线性/非线性预测器的级联形式。我们评估了残差的预测增益和频谱平坦度度量。结果表明:RBF- ar结构是最强大的,EKF训练比RBF网络的标准训练效果更好,非级联RBF- ar预测器的结果优于级联预测器。
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引用次数: 27
期刊
1996 8th European Signal Processing Conference (EUSIPCO 1996)
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