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Iterative identification and restoration of images 图像的迭代识别与恢复
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1990-12-31 DOI: 10.1109/icassp.1988.196759
R. Lagendijk, A. Katsaggelos, J. Biemond
{"title":"Iterative identification and restoration of images","authors":"R. Lagendijk, A. Katsaggelos, J. Biemond","doi":"10.1109/icassp.1988.196759","DOIUrl":"https://doi.org/10.1109/icassp.1988.196759","url":null,"abstract":"The blur identification problem is formulated as a constrained maximum-likelihood problem. The constraints directly incorporate a priori known relations between the blur (and image model) coefficients, such as symmetry properties, into the identification procedure. The resulting nonlinear minimization problem is solved iteratively, yielding a very general identification algorithm. An example of blur identification using synthetic data is given.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1990-12-31","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126736015","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}
引用次数: 155
Adaptive filtering via cumulants and LMS algorithm 通过累积量和LMS算法进行自适应滤波
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-06-01 DOI: 10.1109/ICASSP.1988.196882
Hsing-Hsing Chiang, C. Nikias
{"title":"Adaptive filtering via cumulants and LMS algorithm","authors":"Hsing-Hsing Chiang, C. Nikias","doi":"10.1109/ICASSP.1988.196882","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196882","url":null,"abstract":"A novel adaptive identification scheme is introduced for a nonGaussian white-noise-driven linear, nonminimum-phase FIR (finite-impulse response) system. The adaptive scheme is based on noncausal autoregressive (AR) modeling of higher-order cumulants of the system output. In particular, the magnitude and phase response estimates at each iteration are expressed in terms of the updated parameters of the noncausal AR model. The set of updated AR parameters is obtained by using the LMS (least-mean-squares) algorithm and by using higher-order cumulants instead of time samples of the output signal. It is demonstrated by means of standard examples that the new adaptive scheme works well and, as expected, outperforms the modified (autocorrelation-based) LMS algorithm for nonminimum-phase system identification.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-06-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125234047","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}
引用次数: 11
Automatic detection of image impairments 自动检测图像损伤
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196782
T. D. Toit, J. G. Lourens
{"title":"Automatic detection of image impairments","authors":"T. D. Toit, J. G. Lourens","doi":"10.1109/ICASSP.1988.196782","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196782","url":null,"abstract":"Given only the sampled video image as input, methods to detect automatically image impairments that are visible to the eye are investigated by means of signal processing. The generalized classical cepstral approach is discussed, and thereafter specific investigations of the impairments. Sharpness of focus is measured using rise time properties of edges in the time domain. Echoes or ghosting are identified by investigating the power of the output of a variable correction filter. Top and bottom level signal saturation (also known as white level burning and black level crushing) is detected by grey level histograms.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115296916","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}
引用次数: 2
Impact of noise and encoder/decoder mistracking on ADPCM system performance 噪声和编/解码器误跟踪对ADPCM系统性能的影响
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196559
H. Suyderhoud, S. Dimolitsas
{"title":"Impact of noise and encoder/decoder mistracking on ADPCM system performance","authors":"H. Suyderhoud, S. Dimolitsas","doi":"10.1109/ICASSP.1988.196559","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196559","url":null,"abstract":"An analytic expression is derived that characterizes the distortion occurring in an adaptive differential pulse code modulation (ADPCM) system, when the system encoder and decoder mistrack, in the presence of quantization and coded channel noise. These expressions demonstrate that, when such mistracking occurs, the presence of channel and quantization noise is further accentuated by the transfer function of the mistracked coder system. These results are subsequently extended to the case of coarse quantization where it is shown that in addition to the previous disorder a further term is introduced into the coder transfer function that, depending on the coarseness of the encoder quantization process, can derive the compound system unstable even when the decoder is tracking the encoder.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"57 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115694830","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}
引用次数: 0
A CMOS VLSI cochlea CMOS VLSI耳蜗
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197063
R. Lyon, C. Mead
{"title":"A CMOS VLSI cochlea","authors":"R. Lyon, C. Mead","doi":"10.1109/ICASSP.1988.197063","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197063","url":null,"abstract":"An analog electronic cochlea has been built in CMOS VSLI technology using micropower techniques to achieve the goal of usefulness via realism. The key point of the model and circuits is that a cascade of simple, nearly linear, second-order filter stages with controllable Q parameters suffices to capture the physics of the fluid-dynamic traveling-wave system in the cochlea, including the effects of adaptation and active gain involving the outer hair cells. Measurements on the test chip suggest that the circuit matches both the theory and observations from real cochlea.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115719891","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}
引用次数: 10
Stochastic segment modelling using the estimate-maximize algorithm (speech recognition) 基于估计-最大化算法的随机分段建模(语音识别)
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196528
Salim Roukos, Mari Ostendorf, H. Gish, A. Derr
{"title":"Stochastic segment modelling using the estimate-maximize algorithm (speech recognition)","authors":"Salim Roukos, Mari Ostendorf, H. Gish, A. Derr","doi":"10.1109/ICASSP.1988.196528","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196528","url":null,"abstract":"A probabilistic model called the stochastic segment model is introduced that describes the statistical dependence of all the frames of a speech segment. The model uses a time-warping transformation to map the sequence of observed frames to the appropriate frames of the segment model. The joint density of the observed frames is then given by the joint density of the selected model frames. The automatic training and recognition algorithms are discussed and a few preliminary recognition results are presented.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123124225","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}
引用次数: 22
Adaptive notch filtering in the presence of colored noise 彩色噪声存在时的自适应陷波滤波
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196917
A. Nehorai, P. Stoica
{"title":"Adaptive notch filtering in the presence of colored noise","authors":"A. Nehorai, P. Stoica","doi":"10.1109/ICASSP.1988.196917","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196917","url":null,"abstract":"The authors analyze the convergence of several adaptive notch filter algorithms for sine waves in colored noise, which were recently proposed in the literature. After pointing out the previous algorithms' potential convergence problems, an algorithm is proposed for this problem that does not have convergence problems and provides accurate estimation results.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121789738","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}
引用次数: 7
Robust parametric approach for impulse response estimation 脉冲响应估计的鲁棒参数方法
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/29.7547
U. K. Bhargava, R. Kashyap
{"title":"Robust parametric approach for impulse response estimation","authors":"U. K. Bhargava, R. Kashyap","doi":"10.1109/29.7547","DOIUrl":"https://doi.org/10.1109/29.7547","url":null,"abstract":"The authors consider impulse response estimation of linear time-invariant causal systems based on input-output measurements. In particular, their interest is in developing estimates that are robust against outliers and distributional uncertainties. They present a method which uses Huber's functions as the criterion for fitting a parametric model to the input-output observations. The estimates from this method are compared with the estimates from a similar parametric approach but which uses the conventional quadratic criterion, and also with the estimates from some nonparametric approaches. Results from simulation clearly show that the estimates from the parametric approach using Huber's function are most robust of all the estimates considered.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121808411","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}
引用次数: 8
VLSI structures for computing the Wigner distribution 计算Wigner分布的VLSI结构
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197053
I. Gertner, Moshe Shamash
{"title":"VLSI structures for computing the Wigner distribution","authors":"I. Gertner, Moshe Shamash","doi":"10.1109/ICASSP.1988.197053","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197053","url":null,"abstract":"The authors propose and discuss hybrid VLSI architectures for implementing the Wigner distribution in any dimension. Two contributions are made to the analysis of pipelined architectures for digital signal processing. The first is in introducing the notion of asymptotic area efficiency which in addition to AT/sup 2/ is used as another focal point in the analyses. The second is in presenting a basic design methodology for pipelining arrays under construction of I/O bandwidth by matching their data rate ratios. Well-matched arrays yield good area-time performance with high efficiency. The pipeline methodology forms the basis for new architectures by a systematic construction from arrays that compute subproblems. The authors systematically search for an optimal structure for computing the Wigner distribution in any dimension from arrays that compute subproblems and (static) permutation networks. The pipelining technique yields several structures of which the optimal designs in the sense of AT/sup 2/ and area efficiency are extracted.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"103 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116624855","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}
引用次数: 1
The design of tree-structured M-channel filter banks using perfect reconstruction filter blocks 采用完美重构滤波器块设计树形m通道滤波器组
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196880
E. Viscito, J. Allebach
{"title":"The design of tree-structured M-channel filter banks using perfect reconstruction filter blocks","authors":"E. Viscito, J. Allebach","doi":"10.1109/ICASSP.1988.196880","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196880","url":null,"abstract":"The concept of perfect reconstruction in filter banks is examined using the Smith form of the polyphase matrix. A class of M-channel FIR (finite-impulse response) perfect-reconstruction filter banks is introduced that includes the recently developed lossless filter banks. With the proposed filter banks, the synthesis filters are of the same complexity as the analysis filters. In addition, the synthesis filter bank is easily obtained from the analysis filter bank by inverting a set of M*M constant coefficient matrices. A statistical method for the design of such filter banks is presented. These can be used as building blocks in large tree-structured filter banks where the overall number of channels is any composite integer. An analysis of the computational and storage complexity of such tree structures is given. The results of the analysis are simple formulas for required storage and computation, which helps in the selection of efficient tree structures.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"120960245","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}
引用次数: 17
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