{"title":"Allocation of adaptivity of multistage digital filters","authors":"J. Treichler, M. Larimore, S. Wood","doi":"10.1109/ICASSP.1992.226329","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226329","url":null,"abstract":"Many communications receivers employ a cascade of two or more filters. They are typically designed and implemented separately, each accomplishing its own specific function. Some are fixed in their spectral characteristics, some are selectable, and some, such as data equalizers, are data-adaptive. The incredible growth in the speed and capability of digital signal processing (DSP) devices allows cost-effective digital implementations of many of these filters. This growing digitization would appear to permit the combination of many heretofore separate functions into the same physical filter realization, leading to yet further savings size, weight, power consumption, and cost. The authors examine the case of a digitally implemented tally implemented demodulator which combines the digital filters used for conversion from real sample to complex-valued samples, predecimation filtering, and adaptive equalization. It is shown that it is feasible to combine these functions into a single adaptive filter, but that computational savings are not always attained and that memory requirements almost always grow.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130472506","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}
{"title":"Wake detection: a multichannel approach (SAR image)","authors":"J. Candy","doi":"10.1109/ICASSP.1992.226147","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226147","url":null,"abstract":"The detection of a wake obtained from satellite images using synthetic aperture radar (SAR) techniques is discussed. After both 2-D and 1-D spectral analysis of both simulated and measured wake images, it is concluded that the wake can be considered narrowband temporally and broadband spatially, implying that a plane-wave decomposition may provide a reliable detection approach. In fact, it is shown that the frequency-wavenumber spectrum can be utilized for detection, since the wake decomposition yields plane wave components in symmetric pairs. Various narrowband processors are implemented along with spatial smoothing to provide a reliable frequency-wavenumber estimator. After estimating all of the symmetric wavenumber pairs, various detection schemes are investigated.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"39 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123390252","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}
{"title":"An image processing approach to frequency tracking (application to sonar data)","authors":"J. Abel, H. J. Lee, A. Lowell","doi":"10.1109/ICASSP.1992.225995","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225995","url":null,"abstract":"A method for characterizing a signal's narrowband content is presented. The scenario of interest is one in which the narrowband components are nonstationary, but persist for long periods of time, such as is found in many passive sonar and vibration analysis applications. This so-called frequency tracking problem-generally treated as a one-dimensional problem-is seen to reduce to that of extracting line structures from gray-scale images called spectrograms. Accordingly, the frequency tracking method presented is simply a line extraction technique based on standard image processing and machine vision methods and tuned to the type of line features and noise background present in spectrograms. Initial experiments using real sonar data indicate that the method outperforms currently used methods and has comparable computational cost.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"57 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114333540","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}
Masakatsu Hoshimi, M. Miyata, Shoji Hiraoka, Katsuyuki Niyada
{"title":"Speaker independent speech recognition method using training speech from a small number of speakers","authors":"Masakatsu Hoshimi, M. Miyata, Shoji Hiraoka, Katsuyuki Niyada","doi":"10.1109/ICASSP.1992.225870","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225870","url":null,"abstract":"A novel speaker-independent speech recognition method, which registers speech uttered by a small number of speakers into a dictionary as model speech is presented. It is based on the hypothesis that movement of the vocal tract differs little among individuals when the same word is spoken. This idea leads to the conclusion that dynamic characteristics extracted from a small number of speaker's utterances are effective for speaker-independent speech recognition. A speech recognition method using model utterances in which similarity values of an input word are calculated by matching a small number of speakers' utterances with phoneme templates for speaker-independent recognition is described. When tested with 212 Japanese words, a word recognition rate of 95.8% was obtained. The evaluation of the noise robustness is also reported.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"44 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116353511","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}
{"title":"The equivalence of TLS and correspondence analysis","authors":"K. Yao, F. Lorenzelli, J. Kong","doi":"10.1109/ICASSP.1992.226622","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226622","url":null,"abstract":"The total least squares (TLS) technique is will known in numerical analysis and modern signal processing, while correspondence analysis (CA) has found applications in applied data and clustering analysis. Both techniques use the SVD method to determine their respective optimum solutions. Upon an appropriate preprocessing operation of centering the data matrix, it is shown by minimizing the energy of the perturbation imposed on the data matrix that the two approaches are equivalent. Some simple graphical interpretations and examples are given.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"34 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116360567","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}
{"title":"The fast discrete Radon transform","authors":"B. Kelley, V. Madisetti","doi":"10.1109/ICASSP.1992.226189","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226189","url":null,"abstract":"An explicit relationship between the continuous and discrete time Radon transforms is derived. A generalized least-squares solution to the inversion problem is proposed, and a new inverse counterpart to the fast Radon transform (FRT) algorithm (IFRT) is derived. The authors' interest in the FRT algorithm stems from its application to the seismic inversion problem. A method of seismic migration based upon the wave equation solution in the Radon transform domain is derived using the FRT and IFRT. A number of concurrent VLSI architectures that find favor from the viewpoint of efficient implementation are described. The FRT is extended to three dimensions.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"107 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116362585","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}
{"title":"A technique for defining the architecture and weights of a neural image classifier","authors":"R. Re, F. Roli, S. Serpico, G. Vernazza","doi":"10.1109/ICASSP.1992.226035","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226035","url":null,"abstract":"An approach to setting the architecture and the initial weights of an artificial neural network for solving classification problems is presented. A nonneural phase finds an approximate solution to the classification problems by constraining the shape of classification regions. After an appropriate mapping into a neural net, neural training is applied to refine the solution. Results on an image recognition application are presented.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"29 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121546559","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}
{"title":"Two-tone vs. random process inputs for nonlinear distortion estimation","authors":"Y. S. Cho, E. Powers","doi":"10.1109/ICASSP.1992.226083","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226083","url":null,"abstract":"Two methods (two-tone input approach and random process input approach) for the estimation of harmonic and intermodulation distortion of nonlinear systems are compared. The random input approach, where a nonlinear system is modeled by a second-order Volterra series, is examined in terms of its statistical properties, and its advantages and limitations over the classical two-tone input approach. Experimental results are shown where these two approaches are applied to evaluate second-order distortion of a loudspeaker and to compare the performance of these approaches in terms of Volterra kernels and distortion factors.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"70 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124517484","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}
{"title":"Modified Kalman filtering with an optimal target function","authors":"Liang Li, S. Haykin","doi":"10.1109/ICASSP.1992.226333","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226333","url":null,"abstract":"A general criterion is given to improve the accuracy of the predicted state x(k/k-1) in Kalman filter processing. The criterion is based on the orthogonal relation between the innovations process and past observations. Though this relation is basic to the operation of the Kalman filter, it is often not satisfied in the course of computation because of many target factors. The authors use this relation to construct a target function for minimizing the error. A nonlinear optimal algorithm, combining the standard Kalman filter and the target function equation, is formulated to process the target tracking problem. This algorithm is effective in decreasing the estimation error.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"112 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124070787","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}
{"title":"Invariance of the generalized coherence estimate with respect to reference channel statistics","authors":"D. Sinno, D. Cochran","doi":"10.1109/ICASSP.1992.226009","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226009","url":null,"abstract":"The probability distribution function of the three-channel generalized coherence estimate is shown not to depend on the statistical behavior of the data on one channel, provided the other two channels contain white Gaussian noise and all channels are independent. A technique for multiple-channel matched filtering that is made viable by this invariance result is also discussed.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127552884","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}