Pub Date : 1994-04-19DOI: 10.1109/ICASSP.1994.389851
A. V. Dandawate, G. Giannakis
Extraction of almost periodic signals from their noisy observations is accomplished by exploiting cyclostationarity. The additive noise is allowed to be generally cyclostationary with unknown distribution. Consistency of the proposed estimators is proved and their asymptotic properties are presented. Further, adaptive algorithms are employed for tracking possible time-variations in the parameters of the almost periodic signal. Finally, the proposed methods are tested via simulations.<>
{"title":"Extraction of almost periodic signals using cyclostationarity","authors":"A. V. Dandawate, G. Giannakis","doi":"10.1109/ICASSP.1994.389851","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389851","url":null,"abstract":"Extraction of almost periodic signals from their noisy observations is accomplished by exploiting cyclostationarity. The additive noise is allowed to be generally cyclostationary with unknown distribution. Consistency of the proposed estimators is proved and their asymptotic properties are presented. Further, adaptive algorithms are employed for tracking possible time-variations in the parameters of the almost periodic signal. Finally, the proposed methods are tested via simulations.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"98 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134144805","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.389397
J. H. Kim, Goo-Man Park
For the purpose of digital recording of HDTV signals we have designed a bit rate reduction coder that can reduce the input data rate by a compression ratio greater than 2, while maintaining excellent quality for studio applications. In this paper we have developed an adaptive quantization algorithm to decide scale factor that can optimize bit allocation in a cluster unit. Two-layered code data fixed cluster by cluster serves trick play. This algorithm has been designed to fulfil the constraints of professional studio recorders such as interframe editing, multiple copy, picture quality, trick play and robustness for burst and random errors.<>
{"title":"Two-layered DCT based coding scheme for recording digital HDTV signals","authors":"J. H. Kim, Goo-Man Park","doi":"10.1109/ICASSP.1994.389397","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389397","url":null,"abstract":"For the purpose of digital recording of HDTV signals we have designed a bit rate reduction coder that can reduce the input data rate by a compression ratio greater than 2, while maintaining excellent quality for studio applications. In this paper we have developed an adaptive quantization algorithm to decide scale factor that can optimize bit allocation in a cluster unit. Two-layered code data fixed cluster by cluster serves trick play. This algorithm has been designed to fulfil the constraints of professional studio recorders such as interframe editing, multiple copy, picture quality, trick play and robustness for burst and random errors.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"96 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134237718","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.390092
Yoshihiro Ono, H. Kiya
In this paper, we propose an equivalent model for subband adaptive systems, in order to analyze the performance of the system. Using the proposed model, we obtain a new interpretation of the system, and show theoretical equations of the optimum adaptive digital filter (ADF) coefficients and the least mean squared error (LMSE). From these equations, we also derive the theoretical values of the optimum ADF coefficients and the LMSE. Comparing them with results of computer simulations without the model, we can see agreements of the theoretical values and the simulation results.<>
{"title":"Performance analysis of subband adaptive systems using an equivalent model","authors":"Yoshihiro Ono, H. Kiya","doi":"10.1109/ICASSP.1994.390092","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.390092","url":null,"abstract":"In this paper, we propose an equivalent model for subband adaptive systems, in order to analyze the performance of the system. Using the proposed model, we obtain a new interpretation of the system, and show theoretical equations of the optimum adaptive digital filter (ADF) coefficients and the least mean squared error (LMSE). From these equations, we also derive the theoretical values of the optimum ADF coefficients and the LMSE. Comparing them with results of computer simulations without the model, we can see agreements of the theoretical values and the simulation results.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"23 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134389191","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.389515
T. Saito, T. Komatsu, K. Aizawa
Towards the development of a SHD (super high definition) image acquisition system, previously the authors developed the image-processing based approach with multiple cameras. Originally, in this approach, they used multiple cameras with the same pixel aperture, but in this case there needs to be severe limitations both in the arrangement of multiple cameras and in the configuration of the scene in order to guarantee the spatial uniformity of the resultant resolution. To overcome this difficulty completely, the authors have also previously presented the utilization of multiple cameras with different pixel apertures. The present paper develops a new, alternately iterative image processing algorithm available in the different aperture case. Experimental simulations clearly show that the alternately interactive algorithm behaves satisfactorily.<>
{"title":"An image processing algorithm for a super high definition imaging scheme with multiple different-aperture cameras","authors":"T. Saito, T. Komatsu, K. Aizawa","doi":"10.1109/ICASSP.1994.389515","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389515","url":null,"abstract":"Towards the development of a SHD (super high definition) image acquisition system, previously the authors developed the image-processing based approach with multiple cameras. Originally, in this approach, they used multiple cameras with the same pixel aperture, but in this case there needs to be severe limitations both in the arrangement of multiple cameras and in the configuration of the scene in order to guarantee the spatial uniformity of the resultant resolution. To overcome this difficulty completely, the authors have also previously presented the utilization of multiple cameras with different pixel apertures. The present paper develops a new, alternately iterative image processing algorithm available in the different aperture case. Experimental simulations clearly show that the alternately interactive algorithm behaves satisfactorily.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131492356","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.389499
N. Stacy, M. Burgess, J. Douglass, M. R. Muller, M. Robinson
The INGARA Australian Airborne Radar Surveillance System, formerly called AuSAR, is a low cost airborne imaging radar technology demonstrator under development at the Defence Science and Technology Organisation in Adelaide, Australia. The aims of INGARA are to provide the Australian Defence Forces with a flexible radar system to assess airborne multimode radar technology such as synthetic aperture radar and moving target indicator, to develop in-country expertise in this field, and to provide the scientific and remote sensing community access to an in-country imaging radar sensor for evaluation. The paper describes the real time SAR processor for the INGARA system.<>
{"title":"A real time processor for the Australian synthetic aperture radar","authors":"N. Stacy, M. Burgess, J. Douglass, M. R. Muller, M. Robinson","doi":"10.1109/ICASSP.1994.389499","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389499","url":null,"abstract":"The INGARA Australian Airborne Radar Surveillance System, formerly called AuSAR, is a low cost airborne imaging radar technology demonstrator under development at the Defence Science and Technology Organisation in Adelaide, Australia. The aims of INGARA are to provide the Australian Defence Forces with a flexible radar system to assess airborne multimode radar technology such as synthetic aperture radar and moving target indicator, to develop in-country expertise in this field, and to provide the scientific and remote sensing community access to an in-country imaging radar sensor for evaluation. The paper describes the real time SAR processor for the INGARA system.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131529125","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.390056
J. Perkins, I. Coat
The nonlinear problem of pulse train deinterleaving is mapped into that of line detection in a plane. The image processing technique known as the Hough transform is then applied. The problem is now one of peak detection and the selection of a threshold is examined.<>
{"title":"Pulse train deinterleaving via the Hough transform","authors":"J. Perkins, I. Coat","doi":"10.1109/ICASSP.1994.390056","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.390056","url":null,"abstract":"The nonlinear problem of pulse train deinterleaving is mapped into that of line detection in a plane. The image processing technique known as the Hough transform is then applied. The problem is now one of peak detection and the selection of a threshold is examined.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"31 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131563617","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.389623
H. J. Reekie, J. M. Potter
The paper describes research into compilation techniques for modern, off-the-shelf, floating-point DSP devices. These devices offer a high degree of instruction-level parallelism, which is difficult for compilers to exploit effectively. The authors capture the dataflow and vector nature of DSP programs at the source level, and then focus on the application of standard and novel compilation techniques to utilise this parallelism, especially in critical inner loops. The compiler uses an abstract DSP machine and target machine descriptions to model the special features of modern DSPs. This approach facilitates the development of target-independent code generation algorithms. The authors describe in some detail their loop analysis and code generation algorithms.<>
{"title":"Generating efficient loop code for programmable DSPs","authors":"H. J. Reekie, J. M. Potter","doi":"10.1109/ICASSP.1994.389623","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389623","url":null,"abstract":"The paper describes research into compilation techniques for modern, off-the-shelf, floating-point DSP devices. These devices offer a high degree of instruction-level parallelism, which is difficult for compilers to exploit effectively. The authors capture the dataflow and vector nature of DSP programs at the source level, and then focus on the application of standard and novel compilation techniques to utilise this parallelism, especially in critical inner loops. The compiler uses an abstract DSP machine and target machine descriptions to model the special features of modern DSPs. This approach facilitates the development of target-independent code generation algorithms. The authors describe in some detail their loop analysis and code generation algorithms.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"ii 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131064691","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.389280
F. D. Buø, T. Polzin, A. Waibel
Due to robustness, learnability and ease of integration of different information sources, connectionist parsing systems have proven to be applicable for parsing spoken language, However, most proposed connectionist parsers do not compute and represent complex structures. These parsers assign only a very limited structure to a given input string. For spoken language translation and data base access, more detailed syntactic and semantic representation is needed. In the present paper, the authors show that arbitrary linguistic features and arbitrary complex tree structures can indeed also be learned by a connectionist parsing system.<>
{"title":"Learning complex output representations in connectionist parsing of spoken language","authors":"F. D. Buø, T. Polzin, A. Waibel","doi":"10.1109/ICASSP.1994.389280","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389280","url":null,"abstract":"Due to robustness, learnability and ease of integration of different information sources, connectionist parsing systems have proven to be applicable for parsing spoken language, However, most proposed connectionist parsers do not compute and represent complex structures. These parsers assign only a very limited structure to a given input string. For spoken language translation and data base access, more detailed syntactic and semantic representation is needed. In the present paper, the authors show that arbitrary linguistic features and arbitrary complex tree structures can indeed also be learned by a connectionist parsing system.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":" 41","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132187202","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.389703
E. Harborg, J. E. Knudsen, A. Fuldseth, F. Johansen
In this paper we present an algorithm for the coding of wideband (7 kHz) speech signals at 16 kbps using code excited linear prediction (CELP), primarily intended for use in audio-visual coding systems (e.g., videotelephony) in the ISDN network, or other telecommunication equipment using loudspeaker sound. The algorithm is a full-band approach and much effort has been put into reducing the computational complexity. This has led to a real-time implementation of the present algorithm on a single TMS320C31 DSP (encoder+decoder). Through a formal subjective evaluation we have demonstrated a performance comparable to the CCITT Rec. G.722 subband coder at 64 kbps.<>
{"title":"A real-time wideband CELP coder for a videophone application","authors":"E. Harborg, J. E. Knudsen, A. Fuldseth, F. Johansen","doi":"10.1109/ICASSP.1994.389703","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389703","url":null,"abstract":"In this paper we present an algorithm for the coding of wideband (7 kHz) speech signals at 16 kbps using code excited linear prediction (CELP), primarily intended for use in audio-visual coding systems (e.g., videotelephony) in the ISDN network, or other telecommunication equipment using loudspeaker sound. The algorithm is a full-band approach and much effort has been put into reducing the computational complexity. This has led to a real-time implementation of the present algorithm on a single TMS320C31 DSP (encoder+decoder). Through a formal subjective evaluation we have demonstrated a performance comparable to the CCITT Rec. G.722 subband coder at 64 kbps.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"21 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132697648","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 : 1994-04-19DOI: 10.1109/ICASSP.1994.389666
A. Essebbar
The authors focus on wave parameter estimation. They present a new method using a sliding spatial window along the antenna to improve the performance of parameter estimation and to estimate the shape of the wavefronts. First, ther recall the method which uses a maximum likelihood estimator (MLE) applied on broadband signals. The second part is devoted to applications on real data where the signal to be studied is obtained from a single moving hydrophone measuring the spatial properties of the acoustic multipath propagation.<>
{"title":"Parametric separation applied to underwater acoustic multipath propagation","authors":"A. Essebbar","doi":"10.1109/ICASSP.1994.389666","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389666","url":null,"abstract":"The authors focus on wave parameter estimation. They present a new method using a sliding spatial window along the antenna to improve the performance of parameter estimation and to estimate the shape of the wavefronts. First, ther recall the method which uses a maximum likelihood estimator (MLE) applied on broadband signals. The second part is devoted to applications on real data where the signal to be studied is obtained from a single moving hydrophone measuring the spatial properties of the acoustic multipath propagation.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"32 7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134464375","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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