Pub Date : 2002-12-01DOI: 10.1109/ICASSP.2000.859291
M. Hemmendorff, M. Andersson, T. Kronander, H. Knutsson
We present a method for accurate image registration and motion estimation in multidimensional volumes, such as 3D CT and MR images. The method is based on phase from quadrature filters, which makes it insensitive to variations in luminance and other disturbance in the images. The theory is not restricted to any particular kind of motion model or number of dimensions. Experimental results for affine motions in 3D show high accuracy.
{"title":"Phase-based multidimensional volume registration","authors":"M. Hemmendorff, M. Andersson, T. Kronander, H. Knutsson","doi":"10.1109/ICASSP.2000.859291","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859291","url":null,"abstract":"We present a method for accurate image registration and motion estimation in multidimensional volumes, such as 3D CT and MR images. The method is based on phase from quadrature filters, which makes it insensitive to variations in luminance and other disturbance in the images. The theory is not restricted to any particular kind of motion model or number of dimensions. Experimental results for affine motions in 3D show high accuracy.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133866474","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.861195
A. Austeng, S. Holm
One of the most promising approaches to construct sparse arrays is sparse periodic arrays. The transmitter, Tx, and receiver, Rx, are constructed with different element spacing, both having grating lobes in the radiation pattern occurring at different angles. These are only partially suppressed in the two-way radiation pattern. We present a simple approach to suppress the grating lobes even further. The enhancement is obtained by forcing zeros in the Tx radiation pattern at the occurrence of the Rx grating lobes and vice versa. Weighting of a few of the outer elements has proven sufficient. Enhancement of 30 dB is found for an array with 64 elements aperture with half the elements active. Only the outer group of elements had non-unity weights. The result is comparable to or better than a dense unity weighted array. If desired, additional weighting can be applied.
{"title":"Simple weighting to enhance sparse periodic arrays","authors":"A. Austeng, S. Holm","doi":"10.1109/ICASSP.2000.861195","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861195","url":null,"abstract":"One of the most promising approaches to construct sparse arrays is sparse periodic arrays. The transmitter, Tx, and receiver, Rx, are constructed with different element spacing, both having grating lobes in the radiation pattern occurring at different angles. These are only partially suppressed in the two-way radiation pattern. We present a simple approach to suppress the grating lobes even further. The enhancement is obtained by forcing zeros in the Tx radiation pattern at the occurrence of the Rx grating lobes and vice versa. Weighting of a few of the outer elements has proven sufficient. Enhancement of 30 dB is found for an array with 64 elements aperture with half the elements active. Only the outer group of elements had non-unity weights. The result is comparable to or better than a dense unity weighted array. If desired, additional weighting can be applied.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"46 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115008315","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859289
Sawsan F. Bikhet, A. Darwish, Hany A. Tolba, S. Shaheen
Automated medical image processing and analysis offers a powerful tool for medical diagnosis. In this work we tackle the problem of white blood cell shape analysis based on the morphological characteristics of their outer contour and nuclei. The paper presents a set of preprocessing and segmentation algorithms along with a set of features that are able to recognize and classify different categories of normal white blood cells. The system was tested on gray level images obtained from a CCD camera through a microscope and produced a correct classification rate close to 91%.
{"title":"Segmentation and classification of white blood cells","authors":"Sawsan F. Bikhet, A. Darwish, Hany A. Tolba, S. Shaheen","doi":"10.1109/ICASSP.2000.859289","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859289","url":null,"abstract":"Automated medical image processing and analysis offers a powerful tool for medical diagnosis. In this work we tackle the problem of white blood cell shape analysis based on the morphological characteristics of their outer contour and nuclei. The paper presents a set of preprocessing and segmentation algorithms along with a set of features that are able to recognize and classify different categories of normal white blood cells. The system was tested on gray level images obtained from a CCD camera through a microscope and produced a correct classification rate close to 91%.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"135 1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115368212","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.860091
R. Hegde, Naresh R Shanbhag
We propose a low-power filtering algorithm developed via the soft DSP framework. Soft DSP refers to scaling the supply voltage of a DSP implementation beyond the voltage required to match its critical path delay to the throughput. This deliberate introduction of input-dependent errors leads to degradation in the algorithmic performance, which is then compensated for via algorithmic error-control schemes. The proposed error-control schemes, based on forward/backward linear prediction, provides improved performance over the ones proposed in the past by exploiting correlation in both leading and trailing samples with a latency penalty. It is shown that (a) the proposed scheme provides 60-80% reduction in energy dissipation over that achieved via conventional voltage scaling and (b) for the same algorithmic performance, the overhead involved in the proposed algorithm is more than 50% smaller than existing schemes for medium bandwidth filters.
{"title":"Low-power digital filtering via soft DSP","authors":"R. Hegde, Naresh R Shanbhag","doi":"10.1109/ICASSP.2000.860091","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860091","url":null,"abstract":"We propose a low-power filtering algorithm developed via the soft DSP framework. Soft DSP refers to scaling the supply voltage of a DSP implementation beyond the voltage required to match its critical path delay to the throughput. This deliberate introduction of input-dependent errors leads to degradation in the algorithmic performance, which is then compensated for via algorithmic error-control schemes. The proposed error-control schemes, based on forward/backward linear prediction, provides improved performance over the ones proposed in the past by exploiting correlation in both leading and trailing samples with a latency penalty. It is shown that (a) the proposed scheme provides 60-80% reduction in energy dissipation over that achieved via conventional voltage scaling and (b) for the same algorithmic performance, the overhead involved in the proposed algorithm is more than 50% smaller than existing schemes for medium bandwidth filters.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115414365","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.861809
B. Imperl, Z. Kacic, B. Horvat, A. Zgank
The paper addresses the problem of multilingual acoustic modelling for the design of multilingual speech recognisers. Two different approaches for the definition of multilingual set of triphones (bottom-up and a top-down) are investigated. A new clustering algorithm for the definition of multilingual set of triphones is proposed. The agglomerative clustering algorithm (bottom-up) is based on a definition of a distance measure for triphones defined as a weighted sum of explicit estimates of the context similarity on a monophone level. The monophone similarity estimation method is based on the algorithm of Houtgast. The second type of system uses tree-based clustering (top-down) with a common decision tree. The experiments were based on the SpeechDat II databases (Slovenian, Spanish and German 1000 FDB SpeechDat II). Experiments have shown that the use of the agglomerative clustering algorithm results in a significant reduction of the number of triphones with minor degradation of word accuracy.
{"title":"Agglomerative vs. tree-based clustering for the definition of multilingual set of triphones","authors":"B. Imperl, Z. Kacic, B. Horvat, A. Zgank","doi":"10.1109/ICASSP.2000.861809","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861809","url":null,"abstract":"The paper addresses the problem of multilingual acoustic modelling for the design of multilingual speech recognisers. Two different approaches for the definition of multilingual set of triphones (bottom-up and a top-down) are investigated. A new clustering algorithm for the definition of multilingual set of triphones is proposed. The agglomerative clustering algorithm (bottom-up) is based on a definition of a distance measure for triphones defined as a weighted sum of explicit estimates of the context similarity on a monophone level. The monophone similarity estimation method is based on the algorithm of Houtgast. The second type of system uses tree-based clustering (top-down) with a common decision tree. The experiments were based on the SpeechDat II databases (Slovenian, Spanish and German 1000 FDB SpeechDat II). Experiments have shown that the use of the agglomerative clustering algorithm results in a significant reduction of the number of triphones with minor degradation of word accuracy.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115417784","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.860968
S. Adireddy, L. Tong
The detection of a data sequence with embedded known symbols is considered. For a class of symbol-by-symbol decision feedback receivers, known symbol distributions optimal with respect to the criterion of average mean square error (A-MSE) are presented. Optimal design of the decision feedback receiver is also obtained. Simulation results show that, compared to the performance with conventional symbol placement strategy, considerable gain can be obtained by the joint optimization of symbol placement and equalizer.
{"title":"Detection with embedded known symbols: optimal symbol placement and equalization","authors":"S. Adireddy, L. Tong","doi":"10.1109/ICASSP.2000.860968","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860968","url":null,"abstract":"The detection of a data sequence with embedded known symbols is considered. For a class of symbol-by-symbol decision feedback receivers, known symbol distributions optimal with respect to the criterion of average mean square error (A-MSE) are presented. Optimal design of the decision feedback receiver is also obtained. Simulation results show that, compared to the performance with conventional symbol placement strategy, considerable gain can be obtained by the joint optimization of symbol placement and equalizer.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"54 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123132934","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859058
D. D. Muresan, T. Parks
The detection and estimation of multi-periodic signals of unknown periods in white Gaussian noise is investigated. New estimates for the sub-signals (signals making up the received signal) and their periods are derived using an orthogonal subspace decomposition approach.
{"title":"A new approach to period estimation","authors":"D. D. Muresan, T. Parks","doi":"10.1109/ICASSP.2000.859058","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859058","url":null,"abstract":"The detection and estimation of multi-periodic signals of unknown periods in white Gaussian noise is investigated. New estimates for the sub-signals (signals making up the received signal) and their periods are derived using an orthogonal subspace decomposition approach.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"186 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124405350","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859324
Norbert Strobel, S. Spors, R. Rabenstein
Object localization based on audio and video information is important for the analysis of dynamic scenes, such as video conferences or traffic situations. In this paper, we view the the dynamic audio-video object localization problem as a joint recursive estimation problem. It is solved using a decentralized Kalman filter fusing both audio and video position estimates. To better take into account different object maneuvers, multiple state-space equations are also incorporated. The result is a recursive multi-state multi-sensor estimator. Experiments show that it yields significantly improved joint position estimates compared to results achieved by using either an audio or a video system only.
{"title":"Joint audio-video object localization using a recursive multi-state multi-sensor estimator","authors":"Norbert Strobel, S. Spors, R. Rabenstein","doi":"10.1109/ICASSP.2000.859324","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859324","url":null,"abstract":"Object localization based on audio and video information is important for the analysis of dynamic scenes, such as video conferences or traffic situations. In this paper, we view the the dynamic audio-video object localization problem as a joint recursive estimation problem. It is solved using a decentralized Kalman filter fusing both audio and video position estimates. To better take into account different object maneuvers, multiple state-space equations are also incorporated. The result is a recursive multi-state multi-sensor estimator. Experiments show that it yields significantly improved joint position estimates compared to results achieved by using either an audio or a video system only.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123091073","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.860169
F. Sarigoz, V. B., Vijaya Kumar, J. Bain
We consider the issue of combating signal dropouts in magnetic tape recording. Dropouts are a major limiting factor in the performance of the channel as they cause long bursts of bit errors. To attack this problem, a model of the dropout event has been developed. Based on that model, a strategy has been identified in which an envelope detector followed by a feedforward equalizer is used. The equalization process is done in such a way as to address the three major effects caused by a dropout event: amplitude loss, pulse widening and peakshift. The crucial part of this scheme is the spacing profile estimator, which makes use of an envelope detector. With this scheme, the effects of the dropout are corrected at the expense of a delay in the channel. The model of the dropout and the correction scheme are all presented.
{"title":"Characterization and equalization of dropouts in the magnetic tape recording channel","authors":"F. Sarigoz, V. B., Vijaya Kumar, J. Bain","doi":"10.1109/ICASSP.2000.860169","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860169","url":null,"abstract":"We consider the issue of combating signal dropouts in magnetic tape recording. Dropouts are a major limiting factor in the performance of the channel as they cause long bursts of bit errors. To attack this problem, a model of the dropout event has been developed. Based on that model, a strategy has been identified in which an envelope detector followed by a feedforward equalizer is used. The equalization process is done in such a way as to address the three major effects caused by a dropout event: amplitude loss, pulse widening and peakshift. The crucial part of this scheme is the spacing profile estimator, which makes use of an envelope detector. With this scheme, the effects of the dropout are corrected at the expense of a delay in the channel. The model of the dropout and the correction scheme are all presented.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116778903","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.862012
Jongsun Park, H. Choo, K. Muhammad, Seung Hoon Choi, Yonghee Im, K. Roy
FIR filtering can be expressed as multiplication of a vector by scalars. We present high-speed implementations for adaptive and nonadaptive filters based on a computation sharing multiplier which specifically targets computation re-use in vector-scalar products. The performance of the proposed implementation is compared with implementations based on carry save and Wallace tree multipliers in 0.6 /spl mu/ technology. We show that the sharing multiplier scheme improves speed by approximately 30% and 21% with respect to the Wallace tree multiplier based implementation for non-adaptive and adaptive filters, respectively.
{"title":"Non-adaptive and adaptive filter implementation based on sharing multiplication","authors":"Jongsun Park, H. Choo, K. Muhammad, Seung Hoon Choi, Yonghee Im, K. Roy","doi":"10.1109/ICASSP.2000.862012","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.862012","url":null,"abstract":"FIR filtering can be expressed as multiplication of a vector by scalars. We present high-speed implementations for adaptive and nonadaptive filters based on a computation sharing multiplier which specifically targets computation re-use in vector-scalar products. The performance of the proposed implementation is compared with implementations based on carry save and Wallace tree multipliers in 0.6 /spl mu/ technology. We show that the sharing multiplier scheme improves speed by approximately 30% and 21% with respect to the Wallace tree multiplier based implementation for non-adaptive and adaptive filters, respectively.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116960140","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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