Pub Date : 1995-05-09DOI: 10.1109/ICASSP.1995.479764
J. Maksym
The paper explores the use of an artificial neural network to distinguish between echoes from a constellation of acoustic reflectors representing a target and similar echoes produced by other reflectors, e.g. reverberation. The network was both trained and tested with simulated data. A wide band linear frequency modulated pulse was used in order to resolve the highlights of the target.
{"title":"Neural networks for active echo classification","authors":"J. Maksym","doi":"10.1109/ICASSP.1995.479764","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479764","url":null,"abstract":"The paper explores the use of an artificial neural network to distinguish between echoes from a constellation of acoustic reflectors representing a target and similar echoes produced by other reflectors, e.g. reverberation. The network was both trained and tested with simulated data. A wide band linear frequency modulated pulse was used in order to resolve the highlights of the target.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121755411","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.479444
R. Bamberger, B. Evans, Edward A. Lee, J. McClellan, M. Yoder
A typical path in learning digital signal processing begins at the theoretical end and progresses toward the practical constraints imposed by implementation in hardware or software. On this path, the student would learn how to convert mathematical theory into algorithms and then algorithms into efficient implementations. In this paper, we first summarize the electronic courseware we have already developed in Mathematica, MATLAB, and Ptolemy to teach DSP theory, algorithms, and implementation, respectively. Then, we discuss ways to integrate our efforts to help students discover the connections between these topics.
{"title":"Integrating analysis, simulation, and implementation tools in electronic courseware for teaching signal processing","authors":"R. Bamberger, B. Evans, Edward A. Lee, J. McClellan, M. Yoder","doi":"10.1109/ICASSP.1995.479444","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479444","url":null,"abstract":"A typical path in learning digital signal processing begins at the theoretical end and progresses toward the practical constraints imposed by implementation in hardware or software. On this path, the student would learn how to convert mathematical theory into algorithms and then algorithms into efficient implementations. In this paper, we first summarize the electronic courseware we have already developed in Mathematica, MATLAB, and Ptolemy to teach DSP theory, algorithms, and implementation, respectively. Then, we discuss ways to integrate our efforts to help students discover the connections between these topics.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"49 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121764254","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480115
Agus Trihandoyo, A. Belloum, K. Hou
Achieving reliable implementation of a real-time speaker independent speech recognizer through the telephone network is a challenging research problem. Besides requiring the selection of a suitable algorithm, which takes into account both real-time and accuracy constrains, it requires also adequate hardware architecture and optimized software. This paper presents a dedicated multiprocessor DSP architecture for telecom applications. Based on ADSP-21060 SHARC DSPs, it is the kernel of a digital interactive voice response (IVR) system that is connected to a digital switch through the primary CCITT standard time division multiplexing line of 2.048 Mbps. We attempt to show how a multi-DSP based hardware can be designed for a specific problem in telecommunication, along with the implementation of automatic speech recognition (ASR) to the digital IVR system.
{"title":"A real-time speech recognition architecture for a multi-channel interactive voice response system","authors":"Agus Trihandoyo, A. Belloum, K. Hou","doi":"10.1109/ICASSP.1995.480115","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480115","url":null,"abstract":"Achieving reliable implementation of a real-time speaker independent speech recognizer through the telephone network is a challenging research problem. Besides requiring the selection of a suitable algorithm, which takes into account both real-time and accuracy constrains, it requires also adequate hardware architecture and optimized software. This paper presents a dedicated multiprocessor DSP architecture for telecom applications. Based on ADSP-21060 SHARC DSPs, it is the kernel of a digital interactive voice response (IVR) system that is connected to a digital switch through the primary CCITT standard time division multiplexing line of 2.048 Mbps. We attempt to show how a multi-DSP based hardware can be designed for a specific problem in telecommunication, along with the implementation of automatic speech recognition (ASR) to the digital IVR system.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"105 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115787745","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.479527
K. Lam, O. Au, C. Chan, K. F. Hui, S. F. Lau
The cellular phone is becoming an important means of mobile wireless communication, especially in metropolitan areas. One of the important operating considerations of the cellular phone service providers is the maintainence of the speech quality of the cellular phone network. Subjective evaluation by repeated listening tests at various sites within the coverage area is impractical due to its intrinsic laborious and expensive nature. As a result, it would be much desirable to have an automatic objective evaluation system which applies a good objective speech measure to estimate the statistical average of subjective opinions of the typical conversational speech sentences sent through the cellular network. While extensive work was done for objective speech measures for languages such as English, Japanese, French, and other western languages, little has been done for Chinese. In addition, little has been done to quantify speech quality in the wireless environment.
{"title":"Objective speech measure for Chinese in wireless environment","authors":"K. Lam, O. Au, C. Chan, K. F. Hui, S. F. Lau","doi":"10.1109/ICASSP.1995.479527","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479527","url":null,"abstract":"The cellular phone is becoming an important means of mobile wireless communication, especially in metropolitan areas. One of the important operating considerations of the cellular phone service providers is the maintainence of the speech quality of the cellular phone network. Subjective evaluation by repeated listening tests at various sites within the coverage area is impractical due to its intrinsic laborious and expensive nature. As a result, it would be much desirable to have an automatic objective evaluation system which applies a good objective speech measure to estimate the statistical average of subjective opinions of the typical conversational speech sentences sent through the cellular network. While extensive work was done for objective speech measures for languages such as English, Japanese, French, and other western languages, little has been done for Chinese. In addition, little has been done to quantify speech quality in the wireless environment.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"243 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115950790","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.479520
A. Shen, Benjamim Tang, A. Alwan, G. Pottie
The goal of this study is to develop a robust and high-quality speech coder for wireless communication. The proposed coder is a perceptually-based variable-rate subband coder. The perceptual metric ensures that encoding is optimized to the human listener and is based on calculating the signal-to-mask ratio in short-time frames of the input signal. An adaptive bit allocation scheme is employed and the subband energies are then quantized using a Max-Lloyd quantizer. The coder is fully scalable-increasing the bit rates, improves the quality of encoded speech. Subjective listening tests, using quiet and noisy input signals, indicate that the proposed coder produces high-quality speech when operating at 12 kbps or higher. In error-free conditions, our coder has comparable performance to that of QCELP or GSM coders. For speech in background noise, however, our coder, at 12 kbps, outperforms QCELP significantly, and for music, it outperforms both QCELP and GSM.
{"title":"A robust variable-rate speech coder","authors":"A. Shen, Benjamim Tang, A. Alwan, G. Pottie","doi":"10.1109/ICASSP.1995.479520","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479520","url":null,"abstract":"The goal of this study is to develop a robust and high-quality speech coder for wireless communication. The proposed coder is a perceptually-based variable-rate subband coder. The perceptual metric ensures that encoding is optimized to the human listener and is based on calculating the signal-to-mask ratio in short-time frames of the input signal. An adaptive bit allocation scheme is employed and the subband energies are then quantized using a Max-Lloyd quantizer. The coder is fully scalable-increasing the bit rates, improves the quality of encoded speech. Subjective listening tests, using quiet and noisy input signals, indicate that the proposed coder produces high-quality speech when operating at 12 kbps or higher. In error-free conditions, our coder has comparable performance to that of QCELP or GSM coders. For speech in background noise, however, our coder, at 12 kbps, outperforms QCELP significantly, and for music, it outperforms both QCELP and GSM.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131345304","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.479984
Jae-Gark Choi, Si-Woong Lee, Seong-Dae Kim
This paper presents a segmentation and motion estimation method for object-oriented analysis-synthesis coding. A major difficulty in estimating general motion is that it requires a large area of support in order to achieve a good estimation. Unfortunately, when the supporting area is large it is very likely to have multiple moving objects. To solve this problem, we propose a multi-stage segmentation method which is based on optical flow. The basic concept is to group homogeneous subregions with respect to simpler mapping model into large homogeneous regions with respect to more complex mapping model. By applying a hierarchy of mapping parameter model progressively, we can segment the whole changed region into several parabolic patches. Especially person's face in head-and-shoulder images can be described as one object.
{"title":"Segmentation and motion estimation of moving objects for object-oriented analysis-synthesis coding","authors":"Jae-Gark Choi, Si-Woong Lee, Seong-Dae Kim","doi":"10.1109/ICASSP.1995.479984","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479984","url":null,"abstract":"This paper presents a segmentation and motion estimation method for object-oriented analysis-synthesis coding. A major difficulty in estimating general motion is that it requires a large area of support in order to achieve a good estimation. Unfortunately, when the supporting area is large it is very likely to have multiple moving objects. To solve this problem, we propose a multi-stage segmentation method which is based on optical flow. The basic concept is to group homogeneous subregions with respect to simpler mapping model into large homogeneous regions with respect to more complex mapping model. By applying a hierarchy of mapping parameter model progressively, we can segment the whole changed region into several parabolic patches. Especially person's face in head-and-shoulder images can be described as one object.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"222 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131963128","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480084
J. Spanjaard, L. White
The problem of period uncertainty when evaluating spectrum estimates for wide sense cyclostationary processes is addressed in this paper. In particular, the extended Kalman filter (EKF) and a parallel bank of Kalman filters are investigated as different methods for adaptive estimation of a time-varying period. An example is given concerning an AR(1) process and a number of time-varying periods are adaptively tracked for different periodic functions. Convergence characteristics are also assessed. Finally, a combined detection-estimation approach is also investigated.
{"title":"Adaptive period estimation of a class of periodic random processes","authors":"J. Spanjaard, L. White","doi":"10.1109/ICASSP.1995.480084","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480084","url":null,"abstract":"The problem of period uncertainty when evaluating spectrum estimates for wide sense cyclostationary processes is addressed in this paper. In particular, the extended Kalman filter (EKF) and a parallel bank of Kalman filters are investigated as different methods for adaptive estimation of a time-varying period. An example is given concerning an AR(1) process and a number of time-varying periods are adaptively tracked for different periodic functions. Convergence characteristics are also assessed. Finally, a combined detection-estimation approach is also investigated.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132534151","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480043
R. Buse, Zhi-Qiang Liu
An innovative and powerful method is proposed for measuring physical parameters of lines using the responses from a bank of Gabor (1946) filters. These measurements are made without resorting to an image ruler. First the system is calibrated by establishing a relationship between the frequency of the Gabor filter and line length, then the length and angle of of isolated lines can be measured. A constraint on this method is that the lines in the scene need to be separated and isolated by a minimum distance. Results indicate that Gabor filters can be successfully applied to the measurement of geometric properties of objects, especially where Gabor filters are already being used for processing tasks. The best accuracies in terms of measurement error for the line length and angle measurements were 0.81% and 0.0% respectively.
{"title":"Feature measurement and analysis using Gabor filters","authors":"R. Buse, Zhi-Qiang Liu","doi":"10.1109/ICASSP.1995.480043","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480043","url":null,"abstract":"An innovative and powerful method is proposed for measuring physical parameters of lines using the responses from a bank of Gabor (1946) filters. These measurements are made without resorting to an image ruler. First the system is calibrated by establishing a relationship between the frequency of the Gabor filter and line length, then the length and angle of of isolated lines can be measured. A constraint on this method is that the lines in the scene need to be separated and isolated by a minimum distance. Results indicate that Gabor filters can be successfully applied to the measurement of geometric properties of objects, especially where Gabor filters are already being used for processing tasks. The best accuracies in terms of measurement error for the line length and angle measurements were 0.81% and 0.0% respectively.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130062354","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480566
B. Xuan, R. Bamberger
This paper presents a generalization of the one-dimensional principal component filter bank (PCFB) derived by Tsatsanis (see Univ. of Virginia, Ph.D. Thesis, Sept. 1992.) to higher dimensions. Previously, the results of Tsatsanis were extended to two-dimensional signals, but this was limited to 2D signals and separable resampling operators. The filter bank discussed results in minimizing the mean squared error when only Q out of P subbands are retained. Furthermore, it is shown that the filter bank maximizes the theoretical coding gain (TCG). Simulations are presented, showing the results for reconstructing an image from only the first subband signal, demonstrating the potential of the PCFB.
{"title":"Multi-dimensional, paraunitary principal component filter banks","authors":"B. Xuan, R. Bamberger","doi":"10.1109/ICASSP.1995.480566","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480566","url":null,"abstract":"This paper presents a generalization of the one-dimensional principal component filter bank (PCFB) derived by Tsatsanis (see Univ. of Virginia, Ph.D. Thesis, Sept. 1992.) to higher dimensions. Previously, the results of Tsatsanis were extended to two-dimensional signals, but this was limited to 2D signals and separable resampling operators. The filter bank discussed results in minimizing the mean squared error when only Q out of P subbands are retained. Furthermore, it is shown that the filter bank maximizes the theoretical coding gain (TCG). Simulations are presented, showing the results for reconstructing an image from only the first subband signal, demonstrating the potential of the PCFB.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"162 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134217980","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.479888
K. Bell, Y. Ephraim, H. V. Trees
Bounds on the MSE in estimating the bearing of a planewave signal is of considerable interest in many fields. Of particular importance is the ability of a bound to closely characterize performance in the small error or asymptotic region, and the large error or ambiguity region, and to accurately predict the location of the threshold between the regions. The vector Ziv-Zakai bound is applied to the problem of estimating two-dimensional bearing with planar arrays of arbitrary geometry. The bound is calculated for square and circular arrays, and compared with the Weiss-Weinstein (1983, 1984) bound. The Ziv-Zakai bound is shown to be tighter than the Weiss-Weinstein bound in the threshold and asymptotic regions.
{"title":"Ziv-Zakai lower bounds in bearing estimation","authors":"K. Bell, Y. Ephraim, H. V. Trees","doi":"10.1109/ICASSP.1995.479888","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479888","url":null,"abstract":"Bounds on the MSE in estimating the bearing of a planewave signal is of considerable interest in many fields. Of particular importance is the ability of a bound to closely characterize performance in the small error or asymptotic region, and the large error or ambiguity region, and to accurately predict the location of the threshold between the regions. The vector Ziv-Zakai bound is applied to the problem of estimating two-dimensional bearing with planar arrays of arbitrary geometry. The bound is calculated for square and circular arrays, and compared with the Weiss-Weinstein (1983, 1984) bound. The Ziv-Zakai bound is shown to be tighter than the Weiss-Weinstein bound in the threshold and asymptotic regions.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"37 3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134600488","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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