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2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)最新文献

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m-HDAF multiresolution deformable models m-HDAF多分辨率可变形模型
I. Kakadiaris, M. Papadakis, Lixin Shen, D. Kouri, D. Hoffman
In this paper, we construct a new class of deformable models using new orthogonal wavelets, named modified Hermite distributed approximating functional (m-HDAF) wavelets. The scaling functions of this new family are symmetric and the corresponding wavelets optimize their smoothness for a given number of vanishing moments. In addition, we embed these multiresolution deformable models to the physics-based deformable model framework and use them for fitting 2D and 3D data. We have performed a number of experiments with both synthetic and real data with very encouraging results.
本文利用新的正交小波构造了一类新的可变形模型,即改进的Hermite分布近似泛函(m-HDAF)小波。这个新家族的尺度函数是对称的,相应的小波对给定数量的消失矩优化其平滑性。此外,我们将这些多分辨率可变形模型嵌入到基于物理的可变形模型框架中,并使用它们来拟合2D和3D数据。我们已经用合成数据和真实数据进行了许多实验,结果非常令人鼓舞。
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引用次数: 10
Visual speech recognition using support vector machines 基于支持向量机的视觉语音识别
M. Gordan, Constantine Kotropoulos, I. Pitas
In this paper we propose a visual speech recognition network based on support vector machines. Each word of the dictionary is described as a temporal sequence of visemes. Each viseme is described by a support vector machine, and the temporal character of speech is modeled by integrating the support vector machines as nodes into a Viterbi decoding lattice. Experiments conducted on a small visual speech recognition task show a word recognition rate on the level of the best rates previously reported, even without training the state transition probabilities in the Viterbi lattice and using very simple features. This proves the suitability of support vector machines for visual speech recognition.
本文提出了一种基于支持向量机的视觉语音识别网络。字典中的每个单词都被描述为一个时间序列的词素。用支持向量机描述每个语义,并将支持向量机作为节点集成到维特比解码格中,对语音的时间特征进行建模。在一个小的视觉语音识别任务上进行的实验表明,即使没有训练Viterbi晶格中的状态转移概率并使用非常简单的特征,单词识别率也达到了先前报道的最佳识别率水平。这证明了支持向量机在视觉语音识别中的适用性。
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引用次数: 13
Hierarchic texture classification using statistical steganography techniques 基于统计隐写技术的分层纹理分类
Yu-Kuen Ho, Mei-Yi Wu, Jia-Hong Lee
A novel method for adaptively selecting texture features is presented. We apply statistical steganography techniques with searching for an optimal set of binary masks to extract texture features and provide the best discrimination of texture images. The extracted texture features are robust to noise attacks. Moreover, a tree structure containing the selected set of masks has been set up for classification. Experiments show that the proposed method can achieve high classification rate and also work well in a noise environment.
提出了一种新的纹理特征自适应选择方法。我们应用统计隐写技术,寻找一组最优的二值掩模来提取纹理特征,并提供纹理图像的最佳识别。提取的纹理特征对噪声攻击具有较强的鲁棒性。此外,还建立了包含所选掩码集的树状结构进行分类。实验结果表明,该方法不仅具有较高的分类率,而且在噪声环境下也能很好地工作。
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引用次数: 1
3 classes segmentation for analysis of football audio sequences 用于足球音频序列分析的3类分割
S. Lefèvre, Benjamin Maillard, N. Vincent
We are dealing with segmentation of audio data in order to analyse football audio/video sequences. Audio data is divided into short sequences (typically with duration of one or half a second) which is classified into several classes (speaker, crowd and referee whistle). Every sequence can then be further analysed depending on the class it belongs to. In order to segment audio data, several methods are presented. First simple techniques are reviewed for segmentation in two classes. From the limitations of these approaches, a method based on cepstral analysis is detailed. Next we present two more complex methods dealing with 3 classes segmentation. The first one is based on hidden Markov models whereas the second one is a combination of a C-mean classifier and multidimensional hidden Markov models.
为了分析足球音频/视频序列,我们正在处理音频数据的分割。音频数据被分成短序列(通常持续时间为1秒或半秒),并被分成几类(演讲者、人群和裁判哨声)。每个序列都可以根据它所属的类进一步分析。为了对音频数据进行分割,提出了几种方法。首先回顾了两类分割的简单技术。从这些方法的局限性出发,详细介绍了一种基于倒谱分析的方法。接下来,我们提出两个更复杂的方法处理3类分割。第一种是基于隐马尔可夫模型,第二种是c均值分类器和多维隐马尔可夫模型的结合。
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引用次数: 16
Time series analysis of sunspot oscillations using the wavelet transform 用小波变换分析太阳黑子振荡的时间序列
E. Christopoulou, A. Skodras, A. Georgakilas
By decomposing a time series into time-frequency space, one is able to determine both the dominant mode of variability and how those modes vary in time. We take advantage of this property of the wavelet analysis in order to examine the temporal variation of the period of the umbral oscillations using ground-based observations of the solar atmosphere. We use this real-life signal in order to test the capabilities of different wavelets and to see the problems that arise in analyzing such a signal. We use the continuous wavelet transform in order to perform the analysis and the "a/spl grave/ trous" algorithm as a detrending tool.
通过将时间序列分解为时频空间,人们能够确定可变性的主要模式以及这些模式如何随时间变化。我们利用小波分析的这一特性,利用对太阳大气的地面观测来研究本影振荡周期的时间变化。我们使用这个真实的信号是为了测试不同小波的能力,并看看在分析这样一个信号时出现的问题。我们使用连续小波变换来进行分析,并使用“a/spl grave/ tros”算法作为去趋势工具。
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引用次数: 11
Stochastic approximation and data re-use applied to blind adaptive algorithm for interference cancellation RAKE CDMA receiver 随机逼近和数据复用应用于RAKE CDMA接收机的盲自适应干扰消除算法
Shouhong Zhu, Pushpakanthan Arasaratnam, A. Constantinides
This paper addresses the problem of stochastic approximation and data re-use applied to the blind adaptive algorithm for the interference cancellation RAKE CDMA receiver. The improved adaptive algorithm that utilizes step-size adaptation can achieve both faster convergence and higher steady state performance compared to the fixed step-size ones without increasing much more complexity. Moreover, the improved adaptive algorithm that further utilizes solution averaging and data re-use can further improve both convergence speed and steady state performance. Simulations support the resulting significant improvements.
本文研究了干扰消除RAKE CDMA接收机盲自适应算法中的随机逼近和数据复用问题。采用步长自适应的改进自适应算法在不增加复杂度的前提下,比固定步长自适应算法具有更快的收敛速度和更高的稳态性能。此外,改进的自适应算法进一步利用了解平均和数据重用,进一步提高了收敛速度和稳态性能。仿真结果支持由此产生的显著改进。
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引用次数: 0
A timing recovery criterion derived from the tap weights of a decision feedback equalizer for QAM digital subscriber line systems 基于QAM数字用户线路系统决策反馈均衡器分接权的定时恢复准则
S. Haar, R. Zukunft, F. Vogelbruch
In this paper a timing recovery criterion is derived from the optimum feedforward coefficients of an adaptive symbol-spaced complex-valued decision feedback equalizer (DFE). The criterion is deduced by evaluating the relationship of sampling phase and corresponding, in the minimum mean square error (MMSE) sense optimum, equalizer tap weights exemplarily for a typical digital subscriber line (DSL) scenario. It turns out that a linear combination of the real and imaginary parts of the precursor and cursor coefficient can be applied as timing recovery criterion suitable for digital implementation.
本文从自适应符号间隔复值决策反馈均衡器(DFE)的最优前馈系数出发,导出了时序恢复判据。该准则是通过评估采样相位和相应的最小均方误差(MMSE)感测最优均衡器分接权重的关系推导出来的,以典型数字用户线路(DSL)场景为例。结果表明,前驱体和游标系数的实虚部的线性组合可以作为适合于数字实现的定时恢复判据。
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引用次数: 1
Efficient real-time noise estimation without explicit speech, non-speech detection: an assessment on the AURORA corpus 有效的实时噪声估计没有明确的语音,非语音检测:对AURORA语料库的评估
Nicholas Evans, J. Mason, Benoit G. B. Fauve
This paper addresses the problem of noise estimation for speech enhancement and automatic speech recognition. In the context of mobile telephony, there is a requirement for low resource algorithms which must run at real-time. This paper describes the implementation of a previously published approach, termed quantile-based noise estimation, integrated within a conventional spectral subtraction framework. The novelty lies in the efficiency of the noise estimation process. Assessment is carried out on the AURORA corpus and demonstrates significant improvements in efficiency. Automatic speech recognition results show an average relative improvement of 26% over the baseline.
本文研究了语音增强和自动语音识别中的噪声估计问题。在移动通信环境下,对实时运行的低资源算法有要求。本文描述了一种先前发表的方法的实现,称为基于分位数的噪声估计,集成在传统的谱减法框架中。其新颖之处在于噪声估计过程的效率。对AURORA语料库进行了评估,并显示出效率的显著提高。自动语音识别结果显示平均相对于基线提高了26%。
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引用次数: 8
A prognostic-classification system based on a probabilistic NN for predicting urine bladder cancer recurrence 基于概率神经网络预测膀胱癌复发的预后分类系统
P. Spyridonos, D. Glotsos, D. Cavouras, P. Ravazoula, G. Nikiforidis
In this paper our purpose was to design a prognostic-classification system, based on a probabilistic neural network (PNN), for predicting urine bladder cancer recurrence. Ninety-two patients with bladder cancer were diagnosed and followed up. Images from each patient tissue sample were digitized and an adequate number of nuclei per case were segmented for the generation of morphological and textural nuclear features. Automatic urine bladder tumor characterization as a potential to recur or not was performed utilizing a PNN. An exhaustive search based on classifier performance indicated the best feature combination that produced the minimum classification error. The classification performance of the PNN was optimized employing a 4-dimensional feature vector that comprised one texture feature and three descriptors of nucleus size distribution. The classification accuracy for the group of cases with recurrence was 72.3% (35/47) and 71.1% (32/45) accuracy for the group of cases with no recurrence. The proposed prognostic-system could prove of value in rendering the diagnostic nuclear information a marker of disease recurrence.
在本文中,我们的目的是设计一个基于概率神经网络(PNN)的预测分类系统,用于预测膀胱癌复发。对92例膀胱癌患者进行了诊断和随访。对每个患者组织样本的图像进行数字化处理,并对每个病例中足够数量的细胞核进行分割,以生成形态学和质地核特征。利用PNN自动表征膀胱肿瘤复发或不复发的可能性。基于分类器性能的穷举搜索表明产生最小分类误差的最佳特征组合。采用由一个纹理特征和三个核大小分布描述符组成的四维特征向量对PNN的分类性能进行了优化。复发组的分类准确率为72.3%(35/47),无复发组的分类准确率为71.1%(32/45)。所提出的预后系统在提供诊断性核信息作为疾病复发标记方面具有价值。
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引用次数: 2
Analysis and fast RLS algorithms of quadratic Volterra ADF 二次型Volterra ADF的分析与快速RLS算法
J. Chao
It is shown that adaptive training of quadratic Volterra filters is an ill-conditioned problem, or the error surfaces of the adaptive filters (ADF) are always extremely steep in one particular direction but relatively flat in the rest of the directions. This result is a generalization of a previous report on the special case of when the inputs are delayed values of a single time series of Gaussian distribution. A complete analysis of the correlation matrix of inputs as multiple time series are also obtained for the unrelated case. This paper then presents a fast RLS algorithm for Gaussian input signals costing only O(N/sup 2/) multiplications where N is the number of linear terms in the filter input, the same order as the LMS algorithm, while the RLS algorithm for Volterra ADF costs O(N/sup 5/) multiplications per sample. Simulations shown that this algorithm works well also in non-Gaussian input cases.
结果表明,二次型Volterra滤波器的自适应训练是一个病态问题,即自适应滤波器的误差面在某一特定方向上总是非常陡峭,而在其他方向上则相对平坦。这个结果是对先前关于输入是高斯分布的单个时间序列的延迟值的特殊情况的报告的推广。对于不相关的情况,也得到了输入作为多个时间序列的相关矩阵的完整分析。然后,本文提出了一种用于高斯输入信号的快速RLS算法,只需O(N/sup 2/)次乘法,其中N是滤波器输入中的线性项数,与LMS算法的阶数相同,而用于Volterra ADF的RLS算法每个样本需要O(N/sup 5/)次乘法。仿真结果表明,该算法在非高斯输入情况下也能很好地工作。
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引用次数: 1
期刊
2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)
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