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Design of ARMA digital filters with arbitrary log magnitude function by WLS techniques 利用WLS技术设计任意对数幅值函数的ARMA数字滤波器
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196873
Takao Kobayashi, S. Imai
{"title":"Design of ARMA digital filters with arbitrary log magnitude function by WLS techniques","authors":"Takao Kobayashi, S. Imai","doi":"10.1109/ICASSP.1988.196873","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196873","url":null,"abstract":"A technique is proposed for designing autoregressive-moving average (ARMA) digital filters to have an arbitrary log magnitude frequency response. The technique is based on an iterative weighted least-squares (WLS) approach in the frequency domain. A weight updating procedure is introduced to obtain a least-squares approximation to the given log magnitude function using the WLS approach. Filter coefficients are efficiently calculated using a fast recursive algorithm for a set of linear equations derived from the WLS problem. The technique is also extended to equiripple approximation with a minor modification of the weight updating procedure.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"39 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":"127057056","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
A single chip VLSI architecture for a real time stereo vision processor 一种用于实时立体视觉处理器的单芯片VLSI架构
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197009
F. Jutand, S. Maginot, N. Demassieux, H. Maître
{"title":"A single chip VLSI architecture for a real time stereo vision processor","authors":"F. Jutand, S. Maginot, N. Demassieux, H. Maître","doi":"10.1109/ICASSP.1988.197009","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197009","url":null,"abstract":"The architecture of a single-chip stereo vision processor is presented. It can carry out in real time a dynamic programming algorithm (or a Viterbi algorithm) to compute for each pixel the distance between two corresponding lines. An on-chip surviving-paths memorization and decoding is also described. A rough evaluation for a 1.2- mu m CMOS process provides an area of less than 70 mm/sup 2/.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"78 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":"129926459","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: 3
Optimal pulse shape design using projections onto convex sets 利用凸集上的投影进行最佳脉冲形状设计
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196990
M. Civanlar, R. A. Nobakht
{"title":"Optimal pulse shape design using projections onto convex sets","authors":"M. Civanlar, R. A. Nobakht","doi":"10.1109/ICASSP.1988.196990","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196990","url":null,"abstract":"Recent development in digital signal processing (DSP) hardware has made it possible to use almost any pulse shape for digital data transmission, resulting in a search for design algorithms capable of constructing pulse shapes matching the properties of a given channel. The technique of projection onto convex sets is particularly suitable for the solution of this problem because of its flexibility in modeling a variety of constraints. The use of the technique for the optimal pulse design problem is demonstrated through by an example from power line communications.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"2188 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":"130107075","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
Phoneme recognition: neural networks vs. hidden Markov models vs. hidden Markov models 音素识别:神经网络vs.隐马尔科夫模型vs.隐马尔科夫模型
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196523
A. Waibel, Toshiyuki Hanazawa, Geoffrey E. Hinton, K. Shikano, Kevin J. Lang
{"title":"Phoneme recognition: neural networks vs. hidden Markov models vs. hidden Markov models","authors":"A. Waibel, Toshiyuki Hanazawa, Geoffrey E. Hinton, K. Shikano, Kevin J. Lang","doi":"10.1109/ICASSP.1988.196523","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196523","url":null,"abstract":"A time-delay neural network (TDNN) for phoneme recognition is discussed. By the use of two hidden layers in addition to an input and output layer it is capable of representing complex nonlinear decision surfaces. Three important properties of the TDNNs have been observed. First, it was able to invent without human interference meaningful linguistic abstractions in time and frequency such as formant tracking and segmentation. Second, it has learned to form alternate representations linking different acoustic events with the same higher level concept. In this fashion it can implement trading relations between lower level acoustic events leading to robust recognition performance despite considerable variability in the input speech. Third, the network is translation-invariant and does not rely on precise alignment or segmentation of the input. The TDNNs performance is compared with the best of hidden Markov models (HMMs) on a speaker-dependent phoneme-recognition task. The TDNN achieved a recognition of 98.5% compared to 93.7% for the HMM, i.e., a fourfold reduction in error.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"37 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":"128931846","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: 175
High-speed systolic implementation of fast QR adaptive filters 快速QR自适应滤波器的高速收缩实现
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196912
J. Cioffi
{"title":"High-speed systolic implementation of fast QR adaptive filters","authors":"J. Cioffi","doi":"10.1109/ICASSP.1988.196912","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196912","url":null,"abstract":"A rectangular systolic array of processing units is presented for implementation of the QR adaptive filter. This array requires approximately 8N processing units to exactly solve the least-squares adaptive filtering problem using QR factorization. If a processing unit takes 50 ns to perform a task, the array can be implemented at an adaptive-filter input sampling rate of 20 MHz, with no loss in characteristic high performance (of least squares), numerical stability, or accuracy. This improves on widely used gradient methods for adaptive filtering, which must insert increasing amounts of performance-degrading delay into the adaptive updating when either the speed of implementation or number of taps increase. A discussion of the structure and interconnection of the processing units is included, as well as computer simulations that verify the stability and performance of the adaptive processing array.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"1 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":"129026066","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: 15
Robust detection of object boundaries in Weibull radar imagery 威布尔雷达图像中目标边界的鲁棒检测
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196829
R. A. Brooks, A. Bovik
{"title":"Robust detection of object boundaries in Weibull radar imagery","authors":"R. A. Brooks, A. Bovik","doi":"10.1109/ICASSP.1988.196829","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196829","url":null,"abstract":"Radar image speckle is often modeled as having a negative-exponential, or more generally, gamma distribution. However, studies of noise in coherent radar systems suggest that the first-order statistics may be more accurately modeled using the two-parameter Weibull density, the parameters of which vary with the surface being imaged. Techniques for detecting object boundaries in noisy radar images are proposed and compared. The images are assumed to be coarsely sampled, so that the (multiplicative) radar noise can be modeled as uncorrelated and identically distributed. Edge detection in multiplicative noise is effectively accomplished by thresholding ratios of locally adjacent image estimates. The efficacies of edge detectors defined as ratios of single order statistics, ratios of averages and ratios of best linear unbiased estimators (BLUEs) are compared. The comparisons are based on computed error probabilities as the Weibull parameters are varied. Several example images are provided for empirical comparison as well.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"125 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":"129067466","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
Learning spectral-temporal dependencies using connectionist networks 使用连接网络学习频谱-时间依赖性
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196607
D. Lubensky
{"title":"Learning spectral-temporal dependencies using connectionist networks","authors":"D. Lubensky","doi":"10.1109/ICASSP.1988.196607","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196607","url":null,"abstract":"Describes the application of a layered connectionist network for continuous digit recognition using syllable based segmentation. Knowledge is distributed over many processing units. The behavior of the network in response to a particular input pattern is a collective decision based on the exchange of information among the processing units. A supervised back-propagation learning algorithm is used to repeatedly adjust the weights in the network, to minimize the difference between the actual output vector and the desired output vector. The performance of the network is compared to that of a nearest neighbor classifier trained and tested on the same database. Speaker-dependent continuous digit recognition experiments were performed using a total of 540 digit strings with an average length of 4 digits, collected from six speakers (4 male and 2 female).<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"107 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":"130186487","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: 12
A nesting algorithm for very fast discrete Fourier transforms 快速离散傅里叶变换的嵌套算法
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196997
W. Siu
{"title":"A nesting algorithm for very fast discrete Fourier transforms","authors":"W. Siu","doi":"10.1109/ICASSP.1988.196997","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196997","url":null,"abstract":"The use of a nesting discrete Fourier transform technique to compute discrete Fourier transform is proposed. This technique only relies on two primitive modules and other modules are generated by a standard nesting procedure. The speed of computation of this approach is comparable to the speed of computation of the WFTA, whereas the program size of the present approach is smaller than that of the WFTA. This approach is most suitable for cases where there are restrictions on memory size.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"126 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":"123701317","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
VLSI based design of a battery-operated digital hearing aid 基于VLSI的电池供电数字助听器设计
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197154
R. Morley, G. Engel, T. J. Sullivan, Sundaram M. Natarajan
{"title":"VLSI based design of a battery-operated digital hearing aid","authors":"R. Morley, G. Engel, T. J. Sullivan, Sundaram M. Natarajan","doi":"10.1109/ICASSP.1988.197154","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197154","url":null,"abstract":"A two-chip design is proposed wherein one chip is responsible for data acquisition and reconstruction while a second chip is dedicated to the digital signal-processing circuitry. A custom digital signal processor, potentially capable of performing over 3*10/sup 6/ multiply accumulate operations per second while consuming less than a fraction of a milliwatt, that is required to implement a four-channel hearing aid, is presented. Power consumption is minimized while maintaining a wide dynamic range through the use of sign/logarithm arithmetic.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"06 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":"130562980","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: 19
An adaptive pyramid image coding system 自适应金字塔图像编码系统
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196708
S. Sallent, L. Torres-Urgell
{"title":"An adaptive pyramid image coding system","authors":"S. Sallent, L. Torres-Urgell","doi":"10.1109/ICASSP.1988.196708","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196708","url":null,"abstract":"A pyramidal image structure is presented that uses algorithms defined on arbitrary nonrectangular sampling lattices. An adaptive technique is also developed that assigns to the pyramidal structure an optimum sampling lattice as a function of the frequency content of the original image. Two algorithms, one in the frequency domain and the other in the spatial domain are used to select the optimum sampling lattice.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"50 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":"130697537","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
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