Pub Date : 1900-01-01DOI: 10.1109/ISSPA.2001.950177
H. Al-Mattarneh, Deepak K. Ghodgaonkar, W. Majid
Microwave nondestructive testing methods are fast, contactless, accurate and continuous techniques for the evaluation of moisture content, slope-of-grain, density of knots and specific gravity of timber. Dielectric properties of timber are determined by these factors. We describe how the dielectric properties were deduced from reflection and transmission coefficient measurements of Malaysian timber specimens using a free-space measurement system in the frequency range 8-12.5 GHz. Dielectric properties were evaluated over different angles between the electric field and the direction of grain. Relationships between density, modulus of elasticity, strength, grain angle (angle between the direction of grain and electric field) and dielectric properties of Malaysian timber were established.
{"title":"Microwave nondestructive testing for classification of Malaysian timber using free-space techniques","authors":"H. Al-Mattarneh, Deepak K. Ghodgaonkar, W. Majid","doi":"10.1109/ISSPA.2001.950177","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.950177","url":null,"abstract":"Microwave nondestructive testing methods are fast, contactless, accurate and continuous techniques for the evaluation of moisture content, slope-of-grain, density of knots and specific gravity of timber. Dielectric properties of timber are determined by these factors. We describe how the dielectric properties were deduced from reflection and transmission coefficient measurements of Malaysian timber specimens using a free-space measurement system in the frequency range 8-12.5 GHz. Dielectric properties were evaluated over different angles between the electric field and the direction of grain. Relationships between density, modulus of elasticity, strength, grain angle (angle between the direction of grain and electric field) and dielectric properties of Malaysian timber were established.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122312835","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.949771
B. Dumitrescu, I. Tabus, S. Peltonen, J. Astola, E. Dougherty
The design of an optimal stack filter may be reduced to a linear programming problem, where the criterion is the mean absolute error between a clean signal and noisy version of it. We introduce robustness considerations in the optimization problem. Our approach is to minimize the maximum value of the output distributional influence function (ODIF) associated with the signal and the additive noise. This leads to a new linear programming problem, solved with interior-point methods. We also present a heuristic that allows a significant reduction of the problem size, without sacrificing the quality of the resulting filter.
{"title":"Efficient design of minmax ODIF robust stack filters","authors":"B. Dumitrescu, I. Tabus, S. Peltonen, J. Astola, E. Dougherty","doi":"10.1109/ISSPA.2001.949771","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.949771","url":null,"abstract":"The design of an optimal stack filter may be reduced to a linear programming problem, where the criterion is the mean absolute error between a clean signal and noisy version of it. We introduce robustness considerations in the optimization problem. Our approach is to minimize the maximum value of the output distributional influence function (ODIF) associated with the signal and the additive noise. This leads to a new linear programming problem, solved with interior-point methods. We also present a heuristic that allows a significant reduction of the problem size, without sacrificing the quality of the resulting filter.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127185616","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.950235
S. Bharitkar, C. Kyriakakis
Room acoustical modes, particularly in small rooms, cause a significant variation in the room responses measured at different locations. Responses measured only a few cm apart can vary by up to 15-20 dB at certain frequencies. This makes it difficult to equalize an audio system for multiple simultaneous listeners. Previous methods have utilized multiple microphones and spatial averaging with equal weighting. We determine representative prototypical room responses derived from several room responses that share similar characteristics. We present a fuzzy unsupervised technique for finding similarities between room responses by clustering the room responses, and determining their prototypical responses. These prototypical responses can then be combined to form a general point response. When we use the inverse of the general point response as an equalizing filter, our results show a significant improvement in equalization performance over single point equalization and spatial averaging methods. Our method also allows a unique determination of the optimal number of representatives that are required for a given set of room responses. Applications of this method thus include equalization and multiple point sound control at home and in automobiles.
{"title":"A classification scheme for acoustical room responses","authors":"S. Bharitkar, C. Kyriakakis","doi":"10.1109/ISSPA.2001.950235","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.950235","url":null,"abstract":"Room acoustical modes, particularly in small rooms, cause a significant variation in the room responses measured at different locations. Responses measured only a few cm apart can vary by up to 15-20 dB at certain frequencies. This makes it difficult to equalize an audio system for multiple simultaneous listeners. Previous methods have utilized multiple microphones and spatial averaging with equal weighting. We determine representative prototypical room responses derived from several room responses that share similar characteristics. We present a fuzzy unsupervised technique for finding similarities between room responses by clustering the room responses, and determining their prototypical responses. These prototypical responses can then be combined to form a general point response. When we use the inverse of the general point response as an equalizing filter, our results show a significant improvement in equalization performance over single point equalization and spatial averaging methods. Our method also allows a unique determination of the optimal number of representatives that are required for a given set of room responses. Applications of this method thus include equalization and multiple point sound control at home and in automobiles.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"141 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127325696","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.949845
V. Sucic, B. Boashash
A methodology based on a resolution performance measure P for time-frequency distributions (TFDs) was defined to optimise TFDs and select the one which results into best concentration of signal components around their instantaneous frequency laws, and best suppression of the interference terms in the time-frequency plane. The algorithm which implements this methodology is described by Sucic and Boashash (see Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing-ICASSP'01, Salt Lake City, Utah, USA, May 2001). In this paper we investigate the performance and robustness of the algorithm for multicomponent signals of linear and non-linear FM laws. We also study its applicability to signals with more-than-two components at different distances from each other, as well as the two-component signals with non-equal amplitudes of the components' mainlobes.
定义了一种基于时频分布分辨率性能指标P的方法，以优化时频分布，选择信号分量在其瞬时频率规律附近最集中的时频分布，并在时频平面上最好地抑制干扰项。Sucic和Boashash描述了实现这种方法的算法(参见IEEE声学、语音和信号处理国际会议论文集- icassp '01, Salt Lake City, Utah, USA, 2001年5月)。本文研究了该算法对线性和非线性调频规律的多分量信号的性能和鲁棒性。我们还研究了该方法对两个以上分量的信号在不同距离上的适用性，以及分量主叶幅值不相等的双分量信号。
{"title":"Optimisation algorithm for selecting quadratic time-frequency distributions: performance results and calibration","authors":"V. Sucic, B. Boashash","doi":"10.1109/ISSPA.2001.949845","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.949845","url":null,"abstract":"A methodology based on a resolution performance measure P for time-frequency distributions (TFDs) was defined to optimise TFDs and select the one which results into best concentration of signal components around their instantaneous frequency laws, and best suppression of the interference terms in the time-frequency plane. The algorithm which implements this methodology is described by Sucic and Boashash (see Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing-ICASSP'01, Salt Lake City, Utah, USA, May 2001). In this paper we investigate the performance and robustness of the algorithm for multicomponent signals of linear and non-linear FM laws. We also study its applicability to signals with more-than-two components at different distances from each other, as well as the two-component signals with non-equal amplitudes of the components' mainlobes.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127706396","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.950228
Sun-Hyoung Han, Sung-Woo Kwon, T. Lee, M. Lee
We propose architecture of low power motion estimation algorithm. There are two types of MB (macro block) modes in the proposed algorithm (fast MB mode and normal MB mode). In the fast MB mode, the motion vector found from the previous frame is utilized in the next frame. This mode can be adopted in the conventional fast motion estimation algorithm, and as a result, the computational power is reduced by 40%. In normal the MB mode, among the conventional fast search algorithms, we take the 4SS (four step search), and introduce an additional search method called a center-focused search in the step to increase the PSNR level. We then, implement the corresponding PE (processing element) architecture that gives the hardware performance improvement. The new motion estimation architecture is especially efficient for mobile phone and video conferencing applications in which there is not much motion.
{"title":"Low power motion estimation algorithm based on temporal correlation and its architecture","authors":"Sun-Hyoung Han, Sung-Woo Kwon, T. Lee, M. Lee","doi":"10.1109/ISSPA.2001.950228","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.950228","url":null,"abstract":"We propose architecture of low power motion estimation algorithm. There are two types of MB (macro block) modes in the proposed algorithm (fast MB mode and normal MB mode). In the fast MB mode, the motion vector found from the previous frame is utilized in the next frame. This mode can be adopted in the conventional fast motion estimation algorithm, and as a result, the computational power is reduced by 40%. In normal the MB mode, among the conventional fast search algorithms, we take the 4SS (four step search), and introduce an additional search method called a center-focused search in the step to increase the PSNR level. We then, implement the corresponding PE (processing element) architecture that gives the hardware performance improvement. The new motion estimation architecture is especially efficient for mobile phone and video conferencing applications in which there is not much motion.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"2013 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129853477","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.949778
Wai Yu, R. Ramloll, S. Brewster, B. Riedel
This paper describes the development and evaluation of a haptic interface designed to provide access to line graphs for blind or visually impaired people. Computer-generated line graphs can be felt by users through the sense of touch produced by a PHANToM force feedback device. Experiments have been conducted to test the effectiveness of this interface with both sighted and blind people. The results show that sighted and blind people have achieved about 89.95% and 86.83% correct answers respectively in the experiment.
{"title":"Exploring computer-generated line graphs through virtual touch","authors":"Wai Yu, R. Ramloll, S. Brewster, B. Riedel","doi":"10.1109/ISSPA.2001.949778","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.949778","url":null,"abstract":"This paper describes the development and evaluation of a haptic interface designed to provide access to line graphs for blind or visually impaired people. Computer-generated line graphs can be felt by users through the sense of touch produced by a PHANToM force feedback device. Experiments have been conducted to test the effectiveness of this interface with both sighted and blind people. The results show that sighted and blind people have achieved about 89.95% and 86.83% correct answers respectively in the experiment.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"40 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126359549","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.950186
A. Kountouris, C. Moy, Luc Rambaud, P. Corre
This paper illustrates that the software radio (SWR) approach is of great interest in the context of future multi-standard environments that will become prevalent as we move towards and beyond 3G. Software radio can facilitate this transition thanks to the digital approach that is associated with. The time has come for experimentation and concrete demonstrations that will permit to better understand key issues and establish the merits of the technology. A complete software radio case study is presented here. This case study concerns a real-time triple mode implementation of the GSM-EDGE-UMTS modulation and demodulation schemes using a software centric approach and state-of-the art DSP components. Implementation details and performance results are given. This work together with other efforts around the world demonstrates that SWR will be a key technology factor for the transition from 2G to 3G and the evolution of mobile communications in general.
{"title":"A software radio approach for the transceiver transition from 2G to 2.5G to 3G","authors":"A. Kountouris, C. Moy, Luc Rambaud, P. Corre","doi":"10.1109/ISSPA.2001.950186","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.950186","url":null,"abstract":"This paper illustrates that the software radio (SWR) approach is of great interest in the context of future multi-standard environments that will become prevalent as we move towards and beyond 3G. Software radio can facilitate this transition thanks to the digital approach that is associated with. The time has come for experimentation and concrete demonstrations that will permit to better understand key issues and establish the merits of the technology. A complete software radio case study is presented here. This case study concerns a real-time triple mode implementation of the GSM-EDGE-UMTS modulation and demodulation schemes using a software centric approach and state-of-the art DSP components. Implementation details and performance results are given. This work together with other efforts around the world demonstrates that SWR will be a key technology factor for the transition from 2G to 3G and the evolution of mobile communications in general.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127908859","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.950174
Alexandre Joly, Nathalie Montard, Marcel Buttin
This contribution relates the subjective measurement of audio-visual television quality. There is a need to evaluate the overall continuous quality in audio-visual sequences. The question is to assess the part of audio and video quality and their possible mutual influence. This paper gives the results of a subjective test programme that investigates the interaction between the audio quality and the video quality. The analysis of the database reveals the influence of audio and video on the overall audio-visual quality.
{"title":"Audio-visual quality and interactions between television audio and video","authors":"Alexandre Joly, Nathalie Montard, Marcel Buttin","doi":"10.1109/ISSPA.2001.950174","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.950174","url":null,"abstract":"This contribution relates the subjective measurement of audio-visual television quality. There is a need to evaluate the overall continuous quality in audio-visual sequences. The question is to assess the part of audio and video quality and their possible mutual influence. This paper gives the results of a subjective test programme that investigates the interaction between the audio quality and the video quality. The analysis of the database reveals the influence of audio and video on the overall audio-visual quality.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"90 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117097762","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.950211
L. Stanković
Realization of quadratic and higher order time-frequency (TF) representations based on the short time Fourier transform (STFT) is an practically and theoretically important topic. One possible way to realize quadratic TF distributions by using spectrograms is based on the eigenvalue decomposition of the reduced interference distribution kernel. The other possible method (S-method), which is the topic of this article, is based on the direct relation between the pseudo Wigner distribution (WD) and STFT. In contrast to some other TF representations which are focused on the preservation of marginal properties and the cross terms reduction, this method is derived with the primary goal to preserve the auto-terms quality as in the WD, while avoiding (reducing) the cross-terms. The software and hardware realization of this method is very efficient, since it is completely based on the STFT. By adding one more stage to the already existing systems for the STFT realization one can significantly improve the properties of the TF representation. The S-method can be extended to the cross-terms free (reduced) realizations of the higher order TF representations, such as the polynomial Wigner-Ville distribution, and the L-Wigner distribution. It has also been used for realization of time-scale, and multidimensional space/spatial-frequency representations.
{"title":"Quadratic and higher order time-frequency analysis based on the short-time Fourier transform","authors":"L. Stanković","doi":"10.1109/ISSPA.2001.950211","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.950211","url":null,"abstract":"Realization of quadratic and higher order time-frequency (TF) representations based on the short time Fourier transform (STFT) is an practically and theoretically important topic. One possible way to realize quadratic TF distributions by using spectrograms is based on the eigenvalue decomposition of the reduced interference distribution kernel. The other possible method (S-method), which is the topic of this article, is based on the direct relation between the pseudo Wigner distribution (WD) and STFT. In contrast to some other TF representations which are focused on the preservation of marginal properties and the cross terms reduction, this method is derived with the primary goal to preserve the auto-terms quality as in the WD, while avoiding (reducing) the cross-terms. The software and hardware realization of this method is very efficient, since it is completely based on the STFT. By adding one more stage to the already existing systems for the STFT realization one can significantly improve the properties of the TF representation. The S-method can be extended to the cross-terms free (reduced) realizations of the higher order TF representations, such as the polynomial Wigner-Ville distribution, and the L-Wigner distribution. It has also been used for realization of time-scale, and multidimensional space/spatial-frequency representations.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115318032","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 : 1900-01-01DOI: 10.1109/ISSPA.2001.950190
Yacine Sarni, Rabah Sadoun, A. Belouchrani
The paper presents a novel method of enhancing the interference immunity of a direct sequence spread spectrum system. The method is based on applying a chirp signal to modulate the spreaded sequence. At the receiver we apply a time varying filter to recover the transmitted signal. This scheme is well suited for handling stationary as well as non-stationary interference. In addition, our method improves with respect to the conventional system the receiver performance even when the interference is not present. An analytic expression of the receiver SNR is derived. Simulation results show the improvement in terms of the bit error rate of the proposed technique, with respect to the conventional direct sequence spread spectrum.
{"title":"On the application of chirp modulation in spread spectrum communication systems","authors":"Yacine Sarni, Rabah Sadoun, A. Belouchrani","doi":"10.1109/ISSPA.2001.950190","DOIUrl":"https://doi.org/10.1109/ISSPA.2001.950190","url":null,"abstract":"The paper presents a novel method of enhancing the interference immunity of a direct sequence spread spectrum system. The method is based on applying a chirp signal to modulate the spreaded sequence. At the receiver we apply a time varying filter to recover the transmitted signal. This scheme is well suited for handling stationary as well as non-stationary interference. In addition, our method improves with respect to the conventional system the receiver performance even when the interference is not present. An analytic expression of the receiver SNR is derived. Simulation results show the improvement in terms of the bit error rate of the proposed technique, with respect to the conventional direct sequence spread spectrum.","PeriodicalId":236050,"journal":{"name":"Proceedings of the Sixth International Symposium on Signal Processing and its Applications (Cat.No.01EX467)","volume":"58 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127479165","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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