Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169894
M. V. Malakooti, K. Teague
Because of periodicity of the time series derived from the N angularly equispaced radii, the correlation matrix has an invariant feature under rotation, translation, and scaling. The periodic characteristics possessed by the time series can be utilized to obtain improvement for texture boundary detection. A new circular ARMA (CARMA) model is introduced to represent the time series obtained for shape classification. This model is compared with a regular ARMA model and its high resolution and accuracy is tested for several two dimensional objects. Singular value decomposition (SVD) is used to calculate the insensitive features for shape classification and boundary reconstruction. The invariant right singular vectors of the correlation matrix are used as an orthogonal basis for the solution space. The dimension of the spanned space (model order) is calculated from a new nullity algorithm. To show the high resolution of the eigensystem approach, L1and classical L2solutions are compared.
{"title":"CARMA Model method of two-dimensional shape classification: An eigensystem approach vs. the LP norm","authors":"M. V. Malakooti, K. Teague","doi":"10.1109/ICASSP.1987.1169894","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169894","url":null,"abstract":"Because of periodicity of the time series derived from the N angularly equispaced radii, the correlation matrix has an invariant feature under rotation, translation, and scaling. The periodic characteristics possessed by the time series can be utilized to obtain improvement for texture boundary detection. A new circular ARMA (CARMA) model is introduced to represent the time series obtained for shape classification. This model is compared with a regular ARMA model and its high resolution and accuracy is tested for several two dimensional objects. Singular value decomposition (SVD) is used to calculate the insensitive features for shape classification and boundary reconstruction. The invariant right singular vectors of the correlation matrix are used as an orthogonal basis for the solution space. The dimension of the spanned space (model order) is calculated from a new nullity algorithm. To show the high resolution of the eigensystem approach, L1and classical L2solutions are compared.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125074336","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 : 1987-04-06DOI: 10.1109/ICASSP.1987.1169325
V. Ramamoorthy, T. Natarajan
A conference call can be set up by interconnecting the speech paths in a distributing network. The quality of conference speech is dependent on the circuit conditions and the speech leakage through the hybrids which convert the two wire paths into four wire paths and vice versa. In this paper we analyze the summing algorithm and extend previously published results. We also present a single speaker algorithm that is experimentally shown to be effective.
{"title":"Transmission quality of digital audio teleconferencing bridge","authors":"V. Ramamoorthy, T. Natarajan","doi":"10.1109/ICASSP.1987.1169325","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169325","url":null,"abstract":"A conference call can be set up by interconnecting the speech paths in a distributing network. The quality of conference speech is dependent on the circuit conditions and the speech leakage through the hybrids which convert the two wire paths into four wire paths and vice versa. In this paper we analyze the summing algorithm and extend previously published results. We also present a single speaker algorithm that is experimentally shown to be effective.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122013149","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 : 1987-04-06DOI: 10.1109/ICASSP.1987.1169756
K. Paliwal, A. Basu
In this paper, the problem of speech enhancement when only corrupted speech signal is available for processing is considered. For this, the Kalman filtering method is studied and compared with the Wiener filtering method. Its performance is found to be significantly better than the Wiener filtering method. A delayed-Kalman filtering method is also proposed which improves the speech enhancement performance of Kalman filter further.
{"title":"A speech enhancement method based on Kalman filtering","authors":"K. Paliwal, A. Basu","doi":"10.1109/ICASSP.1987.1169756","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169756","url":null,"abstract":"In this paper, the problem of speech enhancement when only corrupted speech signal is available for processing is considered. For this, the Kalman filtering method is studied and compared with the Wiener filtering method. Its performance is found to be significantly better than the Wiener filtering method. A delayed-Kalman filtering method is also proposed which improves the speech enhancement performance of Kalman filter further.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127702102","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 : 1987-04-06DOI: 10.1109/ICASSP.1987.1169317
I. Pitas, A. Venetsanopoulos
Geophysical interpretation of seismic images is part of geophysical oil prospecting. Its aim is the detection of geologic formations in a seismic cross-section of the earth, which are likely to contain oil reservoirs. It is a labour intensive task and it is heavily based on the experience of the interpreter. Therefore it has not been automated as ithas already been done with other tasks of geophysical seismic signal processing. The aim of this work is to construct an expert system which can automate, at least partly, seismic interpretation. The system developed is able to detect several geologic formations of interest (eg. faults, anticlines, unconformities, salt domes, reefs).
{"title":"AGIS: An expert system for automated geophysical interpretation of seismic images","authors":"I. Pitas, A. Venetsanopoulos","doi":"10.1109/ICASSP.1987.1169317","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169317","url":null,"abstract":"Geophysical interpretation of seismic images is part of geophysical oil prospecting. Its aim is the detection of geologic formations in a seismic cross-section of the earth, which are likely to contain oil reservoirs. It is a labour intensive task and it is heavily based on the experience of the interpreter. Therefore it has not been automated as ithas already been done with other tasks of geophysical seismic signal processing. The aim of this work is to construct an expert system which can automate, at least partly, seismic interpretation. The system developed is able to detect several geologic formations of interest (eg. faults, anticlines, unconformities, salt domes, reefs).","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"12 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130129247","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 : 1987-04-06DOI: 10.1109/ICASSP.1987.1169643
H. Kobatake, Y. Inoue, T. Namai, N. Hamba
A software system for measurement of two-dimensional traffic flow using moving picture processing has been developed. The proposed system is characterized by the functions classifying passing vehicles into one of three classes: motorcycle, small-and large-sized vehicles and detecting their two-dimensional trajectories on a roadway. Experiments to test the system performance have been performed and all of passing vehicles were detected by the system and were classified correctly except for only one large-sized vehicle.
{"title":"Measurement of two-dimensional movement of traffic by image processing","authors":"H. Kobatake, Y. Inoue, T. Namai, N. Hamba","doi":"10.1109/ICASSP.1987.1169643","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169643","url":null,"abstract":"A software system for measurement of two-dimensional traffic flow using moving picture processing has been developed. The proposed system is characterized by the functions classifying passing vehicles into one of three classes: motorcycle, small-and large-sized vehicles and detecting their two-dimensional trajectories on a roadway. Experiments to test the system performance have been performed and all of passing vehicles were detected by the system and were classified correctly except for only one large-sized vehicle.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126539763","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 : 1987-04-06DOI: 10.1109/ICASSP.1987.1169728
R. Mohler, Z. Tang
Affine bilinear time series models AB(p,q) are considered here. An "inverse method" is used to estimate model parameters. A non-anticipative AB(p,p) model could be transfered to 2p-1 dimension vector form AB
{"title":"On identification of non-Gaussian time series","authors":"R. Mohler, Z. Tang","doi":"10.1109/ICASSP.1987.1169728","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169728","url":null,"abstract":"Affine bilinear time series models AB(p,q) are considered here. An \"inverse method\" is used to estimate model parameters. A non-anticipative AB(p,p) model could be transfered to 2p-1 dimension vector form AB","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"78 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127729204","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 : 1987-04-06DOI: 10.1109/ICASSP.1987.1169532
Y. Lim, S. Koh, C. C. Ko
This paper presents a new filtering scheme for splitting speech signals into subbands which results in no aliasing problem when the subband signals are sub-sampled and recombined. The highpass and lowpass filters are an odd-length complementary linear phase filter pair. As a consequence, the highpass filter output can be obtained by subtracting the lowpass filter output from the appropriately delayed input. This produces a factor-of-two reduction in the computation load when compared to the conventional even-length quadrature mirror filter. Another factor-of-two saving in computation load can be gained by using halfband lowpass filters whose every other coefficient is zero. Furthermore, the filters can be designed using well known minimax optimization techniques such as the Remez exchange algorithm and integer programming.
{"title":"Complementary filtering technique for subband speech coder design","authors":"Y. Lim, S. Koh, C. C. Ko","doi":"10.1109/ICASSP.1987.1169532","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169532","url":null,"abstract":"This paper presents a new filtering scheme for splitting speech signals into subbands which results in no aliasing problem when the subband signals are sub-sampled and recombined. The highpass and lowpass filters are an odd-length complementary linear phase filter pair. As a consequence, the highpass filter output can be obtained by subtracting the lowpass filter output from the appropriately delayed input. This produces a factor-of-two reduction in the computation load when compared to the conventional even-length quadrature mirror filter. Another factor-of-two saving in computation load can be gained by using halfband lowpass filters whose every other coefficient is zero. Furthermore, the filters can be designed using well known minimax optimization techniques such as the Remez exchange algorithm and integer programming.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"12 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129201965","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 : 1987-04-01DOI: 10.1109/ICASSP.1987.1169465
M. Amin
The paper provides the conditions on the selection of the lap and the time windows in Wigner-Ville distribution when it is considered for time-varying power spectrum estimation. It is shown that the general class of non-stationary processes requires the inseparability of the two windows. The separation is adequate for a certain class of processes in which the autocorrelation is considered invariant along time intervals whose length is shorter for smaller lags. The paper presents a discussion which relates the time-averaging estimation of the autocorrelation function with the nature of its slow time-variation.
{"title":"Time and lag window selection in Wigner-Ville distribution","authors":"M. Amin","doi":"10.1109/ICASSP.1987.1169465","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169465","url":null,"abstract":"The paper provides the conditions on the selection of the lap and the time windows in Wigner-Ville distribution when it is considered for time-varying power spectrum estimation. It is shown that the general class of non-stationary processes requires the inseparability of the two windows. The separation is adequate for a certain class of processes in which the autocorrelation is considered invariant along time intervals whose length is shorter for smaller lags. The paper presents a discussion which relates the time-averaging estimation of the autocorrelation function with the nature of its slow time-variation.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"31 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114966352","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 : 1987-04-01DOI: 10.1109/ICASSP.1987.1169831
E. Auer
In this paper recursive, second-order, digital filter-structures are given, that are free of zero input limit cycles. This holds for the two cases of sign-magnitude- and two's complement-truncation applied immediately after each multiplication. These structures provide for very inexpensive, but limit cycle free implementations of recursive, digital filters. The results, given for the case of zero input, are in a second part extended to the case of constant input by a simple, so called bypass-structure for the over-all transfer function.
{"title":"Digital filter structures free of limit cycles","authors":"E. Auer","doi":"10.1109/ICASSP.1987.1169831","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169831","url":null,"abstract":"In this paper recursive, second-order, digital filter-structures are given, that are free of zero input limit cycles. This holds for the two cases of sign-magnitude- and two's complement-truncation applied immediately after each multiplication. These structures provide for very inexpensive, but limit cycle free implementations of recursive, digital filters. The results, given for the case of zero input, are in a second part extended to the case of constant input by a simple, so called bypass-structure for the over-all transfer function.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"75 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115269426","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 : 1987-04-01DOI: 10.1109/ICASSP.1987.1169751
D. Borth, Ira A. Gerson, J. Haug
This paper describes the architecture and features of the Motorola DSP56200, an algorithm-specific cascadable digital signal processing peripheral designed to perform the computationally intensive tasks associated with FIR and adaptive FIR digital filtering applications. The DSP56200 is implemented in high performance, low power 1.5µm HCMOS technology and is available in a 28 pin DIP package. The on-chip computation unit includes a 97.5 ns 24×16-bit multiplier with a 40-bit accumulator, a 256×24-bit coefficient RAM, and a 256×16-bit data RAM. Three modes of operation allow the part to be used as a single FIR filter, a dual FIR filter, or a single adaptive FIR filter, with up to 256 taps/chip. In the adaptive FIR filter mode, the part performs the FIR filtering and LMS coefficient update operations for a single tap in 195 ns, permitting use of the part as a 19 kHz sampling rate, 256 tap adaptive FIR filter. Programmable DC tap, coefficient leakage, and adaptation coefficient parameters in the adaptive FIR mode allow the DSP56200 to be used in a wide variety of adaptive FIR filtering applications. The performance of the part in an echo canceller configuration will be presented. Typical applications of the part will also be described.
{"title":"A cascadable adaptive FIR filter VLSI IC","authors":"D. Borth, Ira A. Gerson, J. Haug","doi":"10.1109/ICASSP.1987.1169751","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169751","url":null,"abstract":"This paper describes the architecture and features of the Motorola DSP56200, an algorithm-specific cascadable digital signal processing peripheral designed to perform the computationally intensive tasks associated with FIR and adaptive FIR digital filtering applications. The DSP56200 is implemented in high performance, low power 1.5µm HCMOS technology and is available in a 28 pin DIP package. The on-chip computation unit includes a 97.5 ns 24×16-bit multiplier with a 40-bit accumulator, a 256×24-bit coefficient RAM, and a 256×16-bit data RAM. Three modes of operation allow the part to be used as a single FIR filter, a dual FIR filter, or a single adaptive FIR filter, with up to 256 taps/chip. In the adaptive FIR filter mode, the part performs the FIR filtering and LMS coefficient update operations for a single tap in 195 ns, permitting use of the part as a 19 kHz sampling rate, 256 tap adaptive FIR filter. Programmable DC tap, coefficient leakage, and adaptation coefficient parameters in the adaptive FIR mode allow the DSP56200 to be used in a wide variety of adaptive FIR filtering applications. The performance of the part in an echo canceller configuration will be presented. Typical applications of the part will also be described.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115303834","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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