Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577205
F. Yaghmaee, M. Jamzad
Determining the capacity of watermark for images is a complex task. This capacity may be influenced by many factors such as the content of image and also the strength of watermark. In some recent works the image content has been considered for calculation of watermark capacity. One such approach is to use the noise visibility function (NVF) introduced in (Voloshynovsky, et al., 1999). In this paper we introduced a new method for finding NVF according to the quad tree representation of image instead of using pixel neighborhood data. Our experimental results showed that the depth of a pixel in image quad tree represents much information about the pixel and its maximum allowable distortion compared to pixel neighborhood data as used by the existing NVF computing method. Our method for determining NVF, gave similar results for image capacity compared to the approach based on original NVF calculation as given in (F. Zhang and H. Zhang, June 2004)
确定图像水印的容量是一项复杂的任务。这种能力可能受到图像内容和水印强度等诸多因素的影响。在最近的一些研究中，考虑了图像内容来计算水印容量。其中一种方法是使用(Voloshynovsky, et al.， 1999)中引入的噪声可见性函数(NVF)。本文提出了一种利用图像的四叉树表示代替像素邻域数据寻找NVF的新方法。实验结果表明，与现有NVF计算方法中使用的像素邻域数据相比，图像四叉树中像素的深度代表了像素及其最大允许畸变的更多信息。与基于原始NVF计算的方法(F. Zhang and H. Zhang, 2004年6月)相比，我们确定NVF的方法给出了类似的图像容量结果。
{"title":"Computing watermark capacity in images according to their quad tree","authors":"F. Yaghmaee, M. Jamzad","doi":"10.1109/ISSPIT.2005.1577205","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577205","url":null,"abstract":"Determining the capacity of watermark for images is a complex task. This capacity may be influenced by many factors such as the content of image and also the strength of watermark. In some recent works the image content has been considered for calculation of watermark capacity. One such approach is to use the noise visibility function (NVF) introduced in (Voloshynovsky, et al., 1999). In this paper we introduced a new method for finding NVF according to the quad tree representation of image instead of using pixel neighborhood data. Our experimental results showed that the depth of a pixel in image quad tree represents much information about the pixel and its maximum allowable distortion compared to pixel neighborhood data as used by the existing NVF computing method. Our method for determining NVF, gave similar results for image capacity compared to the approach based on original NVF calculation as given in (F. Zhang and H. Zhang, June 2004)","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"66 9","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"120929092","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577220
N. Rafla
Implementation of a real-time image visualization system on a reconfigurable chip (FPGA) is proposed. The system utilizes an innovative stereoscopic image capture, processing and visualization technique. Implementation is done as a two stage process. In the first stage, the stereo pair is captured using two image sensors. The captured images are then synchronized and sent to the second stage for fusion. A controller module is developed, designed, and placed on the FPGA for this purpose. The second stage is used for reconstruction and visualization of the 3D image. An innovative technique employing dual-processor architecture on the same single FPGA is developed for this purpose. The whole system is placed on a single PCB resulting in a fast processing time and the ability to view 3D video in real-time. The system is simulated, implemented, and tested on real images. Results show that this system is a low cost solution for efficient 3D video visualization using a single chip
{"title":"Real-time 3D image visualization system for digital video on a single chip","authors":"N. Rafla","doi":"10.1109/ISSPIT.2005.1577220","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577220","url":null,"abstract":"Implementation of a real-time image visualization system on a reconfigurable chip (FPGA) is proposed. The system utilizes an innovative stereoscopic image capture, processing and visualization technique. Implementation is done as a two stage process. In the first stage, the stereo pair is captured using two image sensors. The captured images are then synchronized and sent to the second stage for fusion. A controller module is developed, designed, and placed on the FPGA for this purpose. The second stage is used for reconstruction and visualization of the 3D image. An innovative technique employing dual-processor architecture on the same single FPGA is developed for this purpose. The whole system is placed on a single PCB resulting in a fast processing time and the ability to view 3D video in real-time. The system is simulated, implemented, and tested on real images. Results show that this system is a low cost solution for efficient 3D video visualization using a single chip","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123836286","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577110
G. Metikas, S. Olhede
We consider characterisation and analysis of images with dominant local frequencies: this class contains two-dimensional signals written as sums of amplitude and frequency modulated components. We introduce the usage of the multiple vector-valued continuous Morse wavelets for analysis of such signals. We calculate the continuous wavelet transform and the associated scalogram of an image to extract its local amplitude, frequency and dominant orientation. For images contaminated by additive noise, we estimate the local features of the image using several orthogonal mother wavelet functions with optimal localisation: this allows for the construction of estimators of local image features with reduced variability and bias
{"title":"Multiple wavelet analysis of amplitude/frequency modulated images","authors":"G. Metikas, S. Olhede","doi":"10.1109/ISSPIT.2005.1577110","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577110","url":null,"abstract":"We consider characterisation and analysis of images with dominant local frequencies: this class contains two-dimensional signals written as sums of amplitude and frequency modulated components. We introduce the usage of the multiple vector-valued continuous Morse wavelets for analysis of such signals. We calculate the continuous wavelet transform and the associated scalogram of an image to extract its local amplitude, frequency and dominant orientation. For images contaminated by additive noise, we estimate the local features of the image using several orthogonal mother wavelet functions with optimal localisation: this allows for the construction of estimators of local image features with reduced variability and bias","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128037909","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577182
Gang Wang, R. Kraemer
In this paper, a low-complexity approximate channel inverse initialization scheme for blind equalization is proposed. The underlying idea is that the inverse of minimum phase finite impulse response (FIR) channels can be well approximated by inverting the most significant paths of the estimated channel impulse response (CIR). The parameters of the inverse of the truncated channel can be expressed in closed-form expressions in terms of the original CIR and can be used to initialize adaptive equalizers. By applying time-reversal method, the proposed initialization scheme can also be applied for equalization of maximum phase channels. Through extensive computer simulations, we show that channels that cannot be equalized by blind equalizers with the conventional single-spike initialization can now be well equalized with the proposed initialization
{"title":"Low-complexity initialization of adaptive equalizers using approximate channel inverse","authors":"Gang Wang, R. Kraemer","doi":"10.1109/ISSPIT.2005.1577182","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577182","url":null,"abstract":"In this paper, a low-complexity approximate channel inverse initialization scheme for blind equalization is proposed. The underlying idea is that the inverse of minimum phase finite impulse response (FIR) channels can be well approximated by inverting the most significant paths of the estimated channel impulse response (CIR). The parameters of the inverse of the truncated channel can be expressed in closed-form expressions in terms of the original CIR and can be used to initialize adaptive equalizers. By applying time-reversal method, the proposed initialization scheme can also be applied for equalization of maximum phase channels. Through extensive computer simulations, we show that channels that cannot be equalized by blind equalizers with the conventional single-spike initialization can now be well equalized with the proposed initialization","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"100 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115741166","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577187
U. Ali, S.A. Khan
Many efficient wavelet domain estimation algorithms for noise reduction have been proposed in the literature. All of them are based on variance estimation, which require all wavelet coefficients to be saved in the memory thus causing constrains towards the development of real time system. Tarp filter has been used successfully for online/real-time variance estimation of the multilevel wavelets and has shown exceptional performance in image compression. This paper proposes the utilization of the Tarp filter for the estimation of wavelet coefficient variance without saving them in memory. The paper presents the results for Tarp-linear estimator (TLE) and compares it with simple linear estimator (LE). It is found that TLE, while having an advantage of memory free noise reduction method, compromise on SNR
{"title":"Real time Tarp-linear estimator for image noise reduction","authors":"U. Ali, S.A. Khan","doi":"10.1109/ISSPIT.2005.1577187","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577187","url":null,"abstract":"Many efficient wavelet domain estimation algorithms for noise reduction have been proposed in the literature. All of them are based on variance estimation, which require all wavelet coefficients to be saved in the memory thus causing constrains towards the development of real time system. Tarp filter has been used successfully for online/real-time variance estimation of the multilevel wavelets and has shown exceptional performance in image compression. This paper proposes the utilization of the Tarp filter for the estimation of wavelet coefficient variance without saving them in memory. The paper presents the results for Tarp-linear estimator (TLE) and compares it with simple linear estimator (LE). It is found that TLE, while having an advantage of memory free noise reduction method, compromise on SNR","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115878406","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577078
Yuanning Yu, A. Petropulu
We consider the problem of frequency domain identification of a multiple-input multiple-output (MIMO) system driven by white, mutually independent unobservable inputs. In particular, we improve upon a method recently proposed by the authors that uses PARAFAC decomposition of a tensor that is formed based on higher-order statistics of the system output. The approach of Y. Yu and A.P. Petropulu, 2005, utilizes only one slice of the output tensor to recover one row of the system response matrix. We proposed an approach that fully exploits the information in the output tensor, and as a result achieves lower error values. The proposed modification renders the method applicable to systems with more inputs than outputs
{"title":"Improved single PARAFAC decomposition based blind MIMO system estimation","authors":"Yuanning Yu, A. Petropulu","doi":"10.1109/ISSPIT.2005.1577078","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577078","url":null,"abstract":"We consider the problem of frequency domain identification of a multiple-input multiple-output (MIMO) system driven by white, mutually independent unobservable inputs. In particular, we improve upon a method recently proposed by the authors that uses PARAFAC decomposition of a tensor that is formed based on higher-order statistics of the system output. The approach of Y. Yu and A.P. Petropulu, 2005, utilizes only one slice of the output tensor to recover one row of the system response matrix. We proposed an approach that fully exploits the information in the output tensor, and as a result achieves lower error values. The proposed modification renders the method applicable to systems with more inputs than outputs","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115689944","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577105
Jingying Chen, Bernard Paul Tiddeman
Head pose tracking is very important in many human machine interfaces. In this paper, a stereo head pose tracking system is proposed, which detects and tracks six facial features (pupils, nostrils and lip corners) automatically when a human face appears in front of two low-cost USB cameras. The epipolar constraint and a 3D facial feature model are used to improve the accuracy and robustness of the system. This system has the advantages of automatically detecting the facial features and recovering the features lost during the tracking process. Encouraging results have been obtained using the proposed system
{"title":"A real-time stereo head pose tracking system","authors":"Jingying Chen, Bernard Paul Tiddeman","doi":"10.1109/ISSPIT.2005.1577105","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577105","url":null,"abstract":"Head pose tracking is very important in many human machine interfaces. In this paper, a stereo head pose tracking system is proposed, which detects and tracks six facial features (pupils, nostrils and lip corners) automatically when a human face appears in front of two low-cost USB cameras. The epipolar constraint and a 3D facial feature model are used to improve the accuracy and robustness of the system. This system has the advantages of automatically detecting the facial features and recovering the features lost during the tracking process. Encouraging results have been obtained using the proposed system","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"31 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131351711","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577068
L. Golipour, S. Gazor
This paper presents a new approach to an auditory model which matches closely the response patterns in physiological data for single tone inputs. Including the biophysical complexity of the wave motion in the cochlea and considering all terms of the motion equation, the new model can evaluate the auditory spectrum using the Basilar membrane (BM) displacement and the inner hair cells' (IHCs) firing rate for all input signals, including speech. We employ the partial differential motion equations of the BM and its parameters measured for the human auditory system, and design an algorithm which uses short time Fourier transform (STFT) to compute the output for speech stimulus. The idea is to isolate the input signal in the vicinity of a time-window and try to follow the changes in its frequencies and their influences on the signal perceived by the auditory system. The new model includes the nonlinearity action of outer hair cells (OHCs) and provides a new auditory spectrum for speech inputs in the real time domain which reflects a proper view of propagating signal in the cochlea. Despite most of the previous models this model can track the effects of high formant frequencies in the human cochlea as well. This model is a new signal processing tool for studying the response of the auditory system to transient signals which is highly demanded in various speech enhancement and audio coding algorithms
{"title":"A biophysical model of the human cochlea for speech stimulus using STFT","authors":"L. Golipour, S. Gazor","doi":"10.1109/ISSPIT.2005.1577068","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577068","url":null,"abstract":"This paper presents a new approach to an auditory model which matches closely the response patterns in physiological data for single tone inputs. Including the biophysical complexity of the wave motion in the cochlea and considering all terms of the motion equation, the new model can evaluate the auditory spectrum using the Basilar membrane (BM) displacement and the inner hair cells' (IHCs) firing rate for all input signals, including speech. We employ the partial differential motion equations of the BM and its parameters measured for the human auditory system, and design an algorithm which uses short time Fourier transform (STFT) to compute the output for speech stimulus. The idea is to isolate the input signal in the vicinity of a time-window and try to follow the changes in its frequencies and their influences on the signal perceived by the auditory system. The new model includes the nonlinearity action of outer hair cells (OHCs) and provides a new auditory spectrum for speech inputs in the real time domain which reflects a proper view of propagating signal in the cochlea. Despite most of the previous models this model can track the effects of high formant frequencies in the human cochlea as well. This model is a new signal processing tool for studying the response of the auditory system to transient signals which is highly demanded in various speech enhancement and audio coding algorithms","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"141 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133618042","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577080
T. Fujino, J. Nakamura
Recently we have proposed a multi-carrier code division multiple access (MC-CDMA) scheme for the downlink of mobile communications to combat the channel blocking in flat fading multi-input single-output (MISO) channels. In this paper, the previous proposal is extended to the case of frequency-selective fading multi-input multi-output (MIMO) channels. An enhanced scheme is also proposed to obtain better bit error rate (BER) performance and to facilitate signal detection. BER performance of the proposed system using different detection methods is evaluated by numerical calculation. Without decreasing time, frequency, and transmit (Tx) power efficiencies, the proposed system allows us to extract all the Tx data with relatively good BER even when channel blocking is encountered
{"title":"A multicarrier CDMA system for combating channel blocking in frequency-selective fading MIMO channel","authors":"T. Fujino, J. Nakamura","doi":"10.1109/ISSPIT.2005.1577080","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577080","url":null,"abstract":"Recently we have proposed a multi-carrier code division multiple access (MC-CDMA) scheme for the downlink of mobile communications to combat the channel blocking in flat fading multi-input single-output (MISO) channels. In this paper, the previous proposal is extended to the case of frequency-selective fading multi-input multi-output (MIMO) channels. An enhanced scheme is also proposed to obtain better bit error rate (BER) performance and to facilitate signal detection. BER performance of the proposed system using different detection methods is evaluated by numerical calculation. Without decreasing time, frequency, and transmit (Tx) power efficiencies, the proposed system allows us to extract all the Tx data with relatively good BER even when channel blocking is encountered","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132077379","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577087
H. Belkacemi, M. Djeddi, S. Marcos
We consider the problem of joint angle and Doppler estimation for space-time adaptive processing (STAP) airborne radar in non Gaussian clutter modeled as a complex symmetric alpha stable (SaS) process. We introduce a new robust MUSIC algorithm named PFLOM-MUSIC which is obtained using a new class of covariance matrix estimate based on the phased fractional lower-order moment(PFLOM). Such matrix and the subspaces spanned by its eigenvectors serve as a basis for a high-resolution angle/Doppler estimation method. The performance of the PFLOM-MUSIC is examined by comparing the estimation results with those of the sign covariance matrix estimate SCM-MUSIC (T. Liu and J. M. Mendel, Aug. 2001). Extensive Simulations demonstrate that our proposed algorithm outperforms the previously reported estimators
将空时自适应处理(STAP)机载雷达在非高斯杂波环境中的联合角和多普勒估计问题建模为复杂对称α稳定过程。提出了一种新的基于相位分数阶矩(PFLOM)的协方差矩阵估计的鲁棒MUSIC算法PFLOM-MUSIC。该矩阵及其特征向量张成的子空间是高分辨率角度/多普勒估计方法的基础。通过将估计结果与符号协方差矩阵估计SCM-MUSIC的估计结果进行比较，检验了PFLOM-MUSIC的性能(T. Liu and J. M. Mendel, august 2001)。大量的仿真表明，我们提出的算法优于先前报道的估计器
{"title":"Robust subspace technique for joint angle/Doppler estimation in /spl alpha/-stable clutter using phased fractional lower-order moments","authors":"H. Belkacemi, M. Djeddi, S. Marcos","doi":"10.1109/ISSPIT.2005.1577087","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577087","url":null,"abstract":"We consider the problem of joint angle and Doppler estimation for space-time adaptive processing (STAP) airborne radar in non Gaussian clutter modeled as a complex symmetric alpha stable (SaS) process. We introduce a new robust MUSIC algorithm named PFLOM-MUSIC which is obtained using a new class of covariance matrix estimate based on the phased fractional lower-order moment(PFLOM). Such matrix and the subspaces spanned by its eigenvectors serve as a basis for a high-resolution angle/Doppler estimation method. The performance of the PFLOM-MUSIC is examined by comparing the estimation results with those of the sign covariance matrix estimate SCM-MUSIC (T. Liu and J. M. Mendel, Aug. 2001). Extensive Simulations demonstrate that our proposed algorithm outperforms the previously reported estimators","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133869086","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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