Pub Date : 1993-11-01DOI: 10.1109/ACSSC.1993.342498
Bhaskar D. Rao
This paper studies the use of cascade form filter structures for implementing adaptive IIR filters. An efficient scheme for the computation of the gradient is developed. This greatly simplifies the complexity of the adaptive algorithms for cascade implementation of adaptive IIR filters. Issues related to the convergence of the adaptive cascade filter, and computational complexity reduction via alternate mixed form implementations are explored. The application of the cascade IIR adaptive filter for enhancing narrowband/sinusoidal signals in noise is also considered. Simulation results are presented to support the observations.<>
{"title":"Adaptive IIR filtering using cascade structures","authors":"Bhaskar D. Rao","doi":"10.1109/ACSSC.1993.342498","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342498","url":null,"abstract":"This paper studies the use of cascade form filter structures for implementing adaptive IIR filters. An efficient scheme for the computation of the gradient is developed. This greatly simplifies the complexity of the adaptive algorithms for cascade implementation of adaptive IIR filters. Issues related to the convergence of the adaptive cascade filter, and computational complexity reduction via alternate mixed form implementations are explored. The application of the cascade IIR adaptive filter for enhancing narrowband/sinusoidal signals in noise is also considered. Simulation results are presented to support the observations.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"35 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121967764","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342434
J. Gibson, S. P. Stanners, S. McClellan
Variable rate speech coding is well-suited for network and wireless communications and is necessary to maintain good speech quality and intelligibility at ever-lower rates. In 1960 Campbell used a version of the asymptotic equipartition property (AEP) to derive a relationship between the entropy of the source power spectral density and the minimum coefficient rate required to encode the source. We analyze Campbell's coefficient rate expression and investigate its properties for autoregressive (AR) processes and for speech. We compare the coefficient rate to the familiar entropy rate power, and to the AIC model order criterion of Akaike (1971), and consider these quantities as rate indicators for dynamically varying the rate of speech coders.<>
{"title":"Spectral entropy and coefficient rate for speech coding","authors":"J. Gibson, S. P. Stanners, S. McClellan","doi":"10.1109/ACSSC.1993.342434","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342434","url":null,"abstract":"Variable rate speech coding is well-suited for network and wireless communications and is necessary to maintain good speech quality and intelligibility at ever-lower rates. In 1960 Campbell used a version of the asymptotic equipartition property (AEP) to derive a relationship between the entropy of the source power spectral density and the minimum coefficient rate required to encode the source. We analyze Campbell's coefficient rate expression and investigate its properties for autoregressive (AR) processes and for speech. We compare the coefficient rate to the familiar entropy rate power, and to the AIC model order criterion of Akaike (1971), and consider these quantities as rate indicators for dynamically varying the rate of speech coders.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121974298","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342370
M. J. Arnold, M. Roessgen, B. Boashash
A novel technique for filtering a real signal is presented. The method is based upon encoding the signal as an instantaneous frequency (IF) and using IF analysis techniques to recover a filtered version of it. A general iterative time-frequency peak filtering (TFPF) scheme based on this method is formulated. Simulations are given which show extremely clean recovery of signals in noise levels down to -9 dB. Applications to estimating the polynomial phase coefficients of nonstationary signals and differentiating noisy signals are presented.<>
{"title":"Signal filtering using frequency encoding and the time-frequency plane","authors":"M. J. Arnold, M. Roessgen, B. Boashash","doi":"10.1109/ACSSC.1993.342370","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342370","url":null,"abstract":"A novel technique for filtering a real signal is presented. The method is based upon encoding the signal as an instantaneous frequency (IF) and using IF analysis techniques to recover a filtered version of it. A general iterative time-frequency peak filtering (TFPF) scheme based on this method is formulated. Simulations are given which show extremely clean recovery of signals in noise levels down to -9 dB. Applications to estimating the polynomial phase coefficients of nonstationary signals and differentiating noisy signals are presented.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"37 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131923963","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342403
K. Gustafsson, F. McCarthy, A. Paulraj
Calibration errors are the dominant error source for direction finding (DF) from airborne platforms. Calibration of the array manifold is done with the wings in an unknown position. Wing movements during flight perturbs the array manifold from its calibrated value, causing errors in the direction finding. We present a direction finding algorithm that compensates for variations in wing flexure. The algorithm relies on a physically motivated model that captures the gross behavior of the manifold perturbations. The model has been validated using experiments on a model aircraft in an anechoic chamber. The structure of the model can be exploited in estimation schemes such as weighted subspace fitting (WSF), leading to improved DF accuracy. With the correct model parameters, the effect of the manifold perturbation is, in fact, fully neutralized. The properties of the new scheme are demonstrated through analysis and simulation.<>
{"title":"Mitigation of wing flexure for airborne direction-finding applications","authors":"K. Gustafsson, F. McCarthy, A. Paulraj","doi":"10.1109/ACSSC.1993.342403","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342403","url":null,"abstract":"Calibration errors are the dominant error source for direction finding (DF) from airborne platforms. Calibration of the array manifold is done with the wings in an unknown position. Wing movements during flight perturbs the array manifold from its calibrated value, causing errors in the direction finding. We present a direction finding algorithm that compensates for variations in wing flexure. The algorithm relies on a physically motivated model that captures the gross behavior of the manifold perturbations. The model has been validated using experiments on a model aircraft in an anechoic chamber. The structure of the model can be exploited in estimation schemes such as weighted subspace fitting (WSF), leading to improved DF accuracy. With the correct model parameters, the effect of the manifold perturbation is, in fact, fully neutralized. The properties of the new scheme are demonstrated through analysis and simulation.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"224 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132367233","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342417
I. Tabus, D. Petrescu, M. Gabbouj
This paper introduces a training framework for the optimal nonlinear filter design problem. The problem to be solved within the present framework is the selection of the best filter under a data dependent criterion (rather than a model dependent criterion) in one class of nonlinear filters. A class of filters, namely Boolean filters is then considered, for which holds a decoupling property, allowing to transform the initial integer valued problem into the binary domain. The equivalence between the original criterion (in the integer signal domain) and a criterion expressed in the binary signal domain is shown then to hold. The procedures for obtaining the optimal solution for two classes of nonlinear filters, Boolean filters and stack filters, are derived under the new framework. Some numerical simulations are provided, in order to illustrate the effectiveness of the procedures in solving the noise rejection problem.<>
{"title":"Solutions for optimal Boolean and stack filter design under a training framework","authors":"I. Tabus, D. Petrescu, M. Gabbouj","doi":"10.1109/ACSSC.1993.342417","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342417","url":null,"abstract":"This paper introduces a training framework for the optimal nonlinear filter design problem. The problem to be solved within the present framework is the selection of the best filter under a data dependent criterion (rather than a model dependent criterion) in one class of nonlinear filters. A class of filters, namely Boolean filters is then considered, for which holds a decoupling property, allowing to transform the initial integer valued problem into the binary domain. The equivalence between the original criterion (in the integer signal domain) and a criterion expressed in the binary signal domain is shown then to hold. The procedures for obtaining the optimal solution for two classes of nonlinear filters, Boolean filters and stack filters, are derived under the new framework. Some numerical simulations are provided, in order to illustrate the effectiveness of the procedures in solving the noise rejection problem.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130053282","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342545
D.E. Whitehead, G. Coutu, T. Lewis, D. Sturim
A new method is presented of functional mapping of the desired signal used for the training of dynamic supervised neural networks that contain fixed pole IIR structures. The idea is to pass the desired signal through the same number and form of nonlinearities as the data encounters as it passes from the input to the output layer. The neural network has three layers: a filterbank of fixed pole three IIR bandpass filters with variable gains, an intermediate layer of two multiplicative coefficients, and an output layer. The outputs of the input and intermediate layers are passed through logistic nonlinearities.<>
{"title":"Functional mapping of desired signals for improved performance of fully dynamic supervised neural networks with a fixed pole IIR structure","authors":"D.E. Whitehead, G. Coutu, T. Lewis, D. Sturim","doi":"10.1109/ACSSC.1993.342545","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342545","url":null,"abstract":"A new method is presented of functional mapping of the desired signal used for the training of dynamic supervised neural networks that contain fixed pole IIR structures. The idea is to pass the desired signal through the same number and form of nonlinearities as the data encounters as it passes from the input to the output layer. The neural network has three layers: a filterbank of fixed pole three IIR bandpass filters with variable gains, an intermediate layer of two multiplicative coefficients, and an output layer. The outputs of the input and intermediate layers are passed through logistic nonlinearities.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130197906","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342439
P. P. Gandhi, S.S. Rao, R. S. Pappu
We consider a novel baseband waveform coding technique based on wavelets. Wavelets are recognized for their temporal and spectral localization and for their orthogonality across scale and location. We exploit these fundamental properties of wavelets and propose a wavelet-based modulator-demodulator structure to improve communication efficiency. Numerical results for bandwidth occupancy and bandwidth efficiency are given, and a detailed comparison between different families of wavelets is presented.<>
{"title":"On waveform coding using wavelets","authors":"P. P. Gandhi, S.S. Rao, R. S. Pappu","doi":"10.1109/ACSSC.1993.342439","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342439","url":null,"abstract":"We consider a novel baseband waveform coding technique based on wavelets. Wavelets are recognized for their temporal and spectral localization and for their orthogonality across scale and location. We exploit these fundamental properties of wavelets and propose a wavelet-based modulator-demodulator structure to improve communication efficiency. Numerical results for bandwidth occupancy and bandwidth efficiency are given, and a detailed comparison between different families of wavelets is presented.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"2009 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131526394","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342305
Jing Fang, L. Atlas
Quadratic detectors have been shown to have good frequency selectivity, short transient response, and effective noise suppression. We demonstrate how phase shifts in phase-shift-keyed signals are easily detected even in strong background noise, and how short-duration frequency variation can be accurately evaluated. Moreover design constraints of quadratic detectors are discussed and trade-offs are presented. Our simulations showed that the quadratic detector provides a simple method for the detection of phase shifts and frequency shifts in clean or noisy environments. These detectors and the design technique also have general signal processing applications.<>
{"title":"Quadratic detectors for noise-immune detection of modulated signals","authors":"Jing Fang, L. Atlas","doi":"10.1109/ACSSC.1993.342305","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342305","url":null,"abstract":"Quadratic detectors have been shown to have good frequency selectivity, short transient response, and effective noise suppression. We demonstrate how phase shifts in phase-shift-keyed signals are easily detected even in strong background noise, and how short-duration frequency variation can be accurately evaluated. Moreover design constraints of quadratic detectors are discussed and trade-offs are presented. Our simulations showed that the quadratic detector provides a simple method for the detection of phase shifts and frequency shifts in clean or noisy environments. These detectors and the design technique also have general signal processing applications.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"44 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131601811","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342323
B. Brower, D. Couwenhoven, B. Gandhi, C. Smith
Under funding from the Defense LANDSAT Program Office, a method was developed for compressing multispectral data. A rate-controlled adaptive differential pulse code modulation technique was developed with minimal complexity for downlink applications. This algorithm uses an adaptive 2-D and 3-D prediction of pixels. The difference between the predicted and original pixel is quantized with a locally adaptive quantizer. This technique has produced compression ratios of up to 2.5:1 losslessly and up to 5:1 with minimal visual image quality loss on LANDSAT and M-7 high resolution multispectral data. This paper describes the ADPCM algorithm and its impact on numerical, visual and machine exploitation performance.<>
{"title":"ADPCM for advanced LANDSAT downlink applications","authors":"B. Brower, D. Couwenhoven, B. Gandhi, C. Smith","doi":"10.1109/ACSSC.1993.342323","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342323","url":null,"abstract":"Under funding from the Defense LANDSAT Program Office, a method was developed for compressing multispectral data. A rate-controlled adaptive differential pulse code modulation technique was developed with minimal complexity for downlink applications. This algorithm uses an adaptive 2-D and 3-D prediction of pixels. The difference between the predicted and original pixel is quantized with a locally adaptive quantizer. This technique has produced compression ratios of up to 2.5:1 losslessly and up to 5:1 with minimal visual image quality loss on LANDSAT and M-7 high resolution multispectral data. This paper describes the ADPCM algorithm and its impact on numerical, visual and machine exploitation performance.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132623402","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 : 1993-11-01DOI: 10.1109/ACSSC.1993.342572
Christopher F. Barnes, E. J. Holder
A novel combination of a successive approximation scalar quantizer encoder structure and a direct sum decoder structure is introduced. The encoder structure is a multiple level free structure that corresponds to a conventional uniform scalar quantizer. The decoder structure is a multiple stage direct sum structure. The uniform quantizer encoder structure is matched to the dynamic range of the source output. The decoder direct sum structure is jointly optimized for the source probability density function and the fixed partition of the uniform quantizer encoder. The bit stream generated by the encoder can be decoded either by the standard uniform quantizer decoder, or by the generalized nonuniform direct sum decoder described. Successive approximation multiple stage scalar quantizers with both standard and generalized decoders are tested on image wavelet transform coefficients. The generalized codes give approximately 0/spl sim/7 dB improvements over standard codes for one to eight bit representations of real-valued wavelet coefficients.<>
{"title":"Successive approximation quantization with generalized decoding for wavelet transform image coding","authors":"Christopher F. Barnes, E. J. Holder","doi":"10.1109/ACSSC.1993.342572","DOIUrl":"https://doi.org/10.1109/ACSSC.1993.342572","url":null,"abstract":"A novel combination of a successive approximation scalar quantizer encoder structure and a direct sum decoder structure is introduced. The encoder structure is a multiple level free structure that corresponds to a conventional uniform scalar quantizer. The decoder structure is a multiple stage direct sum structure. The uniform quantizer encoder structure is matched to the dynamic range of the source output. The decoder direct sum structure is jointly optimized for the source probability density function and the fixed partition of the uniform quantizer encoder. The bit stream generated by the encoder can be decoded either by the standard uniform quantizer decoder, or by the generalized nonuniform direct sum decoder described. Successive approximation multiple stage scalar quantizers with both standard and generalized decoders are tested on image wavelet transform coefficients. The generalized codes give approximately 0/spl sim/7 dB improvements over standard codes for one to eight bit representations of real-valued wavelet coefficients.<<ETX>>","PeriodicalId":266447,"journal":{"name":"Proceedings of 27th Asilomar Conference on Signals, Systems and Computers","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131103983","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}
京公网安备 11010802042870号
京ICP备2023020795号-1
扫码关注我们
求助内容：
应助结果提醒方式：