Pub Date : 1996-09-01DOI: 10.1109/DSPWS.1996.555536
V. Lukin, T. Saramaki
Step-like weighting windows are proposed for various applications in digital signal processing. Their parameters are optimized to minimize either the maximal sidelobe level of the corresponding frequency response (equidistant array pattern) or the ratio of the energies concentrated in the sidelobes and in the mainlobe. These parameters together with the mainlobe width are compared to the corresponding values of the optimal Dolph-Chebyshev and Kaiser-Bessel responses. Several advantages provided by the step-like window implementation are considered.
{"title":"Step-like weighting windows for DSP in spectral analysis and antenna arrays","authors":"V. Lukin, T. Saramaki","doi":"10.1109/DSPWS.1996.555536","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555536","url":null,"abstract":"Step-like weighting windows are proposed for various applications in digital signal processing. Their parameters are optimized to minimize either the maximal sidelobe level of the corresponding frequency response (equidistant array pattern) or the ratio of the energies concentrated in the sidelobes and in the mainlobe. These parameters together with the mainlobe width are compared to the corresponding values of the optimal Dolph-Chebyshev and Kaiser-Bessel responses. Several advantages provided by the step-like window implementation are considered.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132114946","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555516
H. Skinnemoen
Vector quantization (VQ) is a multidimensional block quantizer methodology that can be very efficient with respect to approaching the rate-distortion bounds. Best performance is obtained for longer blocks. However, as the vectors become longer, the number of codebook vectors will generally increase, and the complexity of searching the codebook for the best codebook vector may soon become prohibitive. For best quantizer performance, the VQ must be trained for the source, but this usually prohibits the use of structured (or algebraic) codebooks that are fast to search. This paper presents a novel methodology for codebook design that combines the traditional training of codebooks by the well proven generalized Lloyd algorithm (GLA) with a structured codebook that can be searched efficiently. The concept is termed gradient search algorithm (GSA) since it is based upon a gradient in the error surface of the codebook pointing towards the optimum codebook vector choice.
{"title":"A codebook design method for fast VQ search","authors":"H. Skinnemoen","doi":"10.1109/DSPWS.1996.555516","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555516","url":null,"abstract":"Vector quantization (VQ) is a multidimensional block quantizer methodology that can be very efficient with respect to approaching the rate-distortion bounds. Best performance is obtained for longer blocks. However, as the vectors become longer, the number of codebook vectors will generally increase, and the complexity of searching the codebook for the best codebook vector may soon become prohibitive. For best quantizer performance, the VQ must be trained for the source, but this usually prohibits the use of structured (or algebraic) codebooks that are fast to search. This paper presents a novel methodology for codebook design that combines the traditional training of codebooks by the well proven generalized Lloyd algorithm (GLA) with a structured codebook that can be searched efficiently. The concept is termed gradient search algorithm (GSA) since it is based upon a gradient in the error surface of the codebook pointing towards the optimum codebook vector choice.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129842442","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555560
P. Grohan, S. Marcos
The paper revisits the problem of equalization with an IIR filter. The use of the discrete Kalman observer is adopted to perform the IIR equalization and to circumvent the minimum phase spectral factorization. A unified study of adaptive equalizers based on the Kalman observer is presented, a novel adaptive Kalman equalizer scheme is discussed and finally, a performance comparison is given by computer simulations.
{"title":"Structures and performances of several adaptive Kalman equalizers","authors":"P. Grohan, S. Marcos","doi":"10.1109/DSPWS.1996.555560","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555560","url":null,"abstract":"The paper revisits the problem of equalization with an IIR filter. The use of the discrete Kalman observer is adopted to perform the IIR equalization and to circumvent the minimum phase spectral factorization. A unified study of adaptive equalizers based on the Kalman observer is presented, a novel adaptive Kalman equalizer scheme is discussed and finally, a performance comparison is given by computer simulations.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"49 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124845417","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555555
D. Stanhill, Y. Zeevi
The discrete wavelet transform, and multiresolution analysis, can be viewed as the application of a non-uniform filter bank. The polyphase matrix representation of the corresponding frame operator is given for general oversampled wavelet-type filter banks. The issue of lower and upper bounds of the frame and their relation to those of the underlying one level filter bank frame is treated, and bounds are given in a few special cases. A few examples are presented demonstrating the use of the results obtained.
{"title":"Frame analysis of wavelet type filter banks","authors":"D. Stanhill, Y. Zeevi","doi":"10.1109/DSPWS.1996.555555","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555555","url":null,"abstract":"The discrete wavelet transform, and multiresolution analysis, can be viewed as the application of a non-uniform filter bank. The polyphase matrix representation of the corresponding frame operator is given for general oversampled wavelet-type filter banks. The issue of lower and upper bounds of the frame and their relation to those of the underlying one level filter bank frame is treated, and bounds are given in a few special cases. A few examples are presented demonstrating the use of the results obtained.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"222 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123477037","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555457
I. Balasingham, A. Fuldseth, T. Ramstad
Optimized perfect reconstruction octave-band two-stage and three-stage tree-structured filter banks and optimized uniform four-channel and eight-channel filter banks are compared for hybrid video coding. The filter coefficients are optimized at each stage for subband coding gain for nonuniform filter banks. Six types of frequency partitionings are presented to compare the efficient utilization of the spatial redundancies in the prediction error image. The performance of the proposed filter banks are compared for PSNR and visual quality. The overall performance indicates that the optimized perfect reconstruction octave-band two-stage tree-structured (FB 22_22_8) filter bank is a good candidate for video coding.
{"title":"A comparison of different optimized structures of filter banks for video coding","authors":"I. Balasingham, A. Fuldseth, T. Ramstad","doi":"10.1109/DSPWS.1996.555457","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555457","url":null,"abstract":"Optimized perfect reconstruction octave-band two-stage and three-stage tree-structured filter banks and optimized uniform four-channel and eight-channel filter banks are compared for hybrid video coding. The filter coefficients are optimized at each stage for subband coding gain for nonuniform filter banks. Six types of frequency partitionings are presented to compare the efficient utilization of the spatial redundancies in the prediction error image. The performance of the proposed filter banks are compared for PSNR and visual quality. The overall performance indicates that the optimized perfect reconstruction octave-band two-stage tree-structured (FB 22_22_8) filter bank is a good candidate for video coding.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"34 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128210742","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555567
T. Falk, M. Gotz, T. Kilias, W. Schwarz, T. Ruhlicke
We present the principle and a realisation of an integrated circuit for the generation of time discrete broadband signals. The principle of the circuit is based on deterministic nonlinear systems using Markov maps as the system function. The generated signals behave chaotically. The probability density function, the auto-correlation coefficients and also the power density function of the signal depend on the characteristics of the system map and can be adjusted digitally by the change of its coefficients. The circuit consists of an analogue part which processes the signals and a digital one for storing the parameters of the system. Because the signal generating system is an analogue one the generated signals have an infinite period length. Applications can be found in speech generation, measuring technologies and simulating noise in telecommunication channels.
{"title":"A chaos-based programmable analogue-digital circuit for broadband signal generation","authors":"T. Falk, M. Gotz, T. Kilias, W. Schwarz, T. Ruhlicke","doi":"10.1109/DSPWS.1996.555567","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555567","url":null,"abstract":"We present the principle and a realisation of an integrated circuit for the generation of time discrete broadband signals. The principle of the circuit is based on deterministic nonlinear systems using Markov maps as the system function. The generated signals behave chaotically. The probability density function, the auto-correlation coefficients and also the power density function of the signal depend on the characteristics of the system map and can be adjusted digitally by the change of its coefficients. The circuit consists of an analogue part which processes the signals and a digital one for storing the parameters of the system. Because the signal generating system is an analogue one the generated signals have an infinite period length. Applications can be found in speech generation, measuring technologies and simulating noise in telecommunication channels.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124142433","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555570
T. Leung, Pr White
This paper discusses a fuzzy logic based system developed to detect and classify a series of real life underwater acoustic transients. Signals received in the ocean are highly contaminated by various sources. While a trained sonar operator may classify transients without too difficulty, automatic classification by traditional template matching does not give satisfactory results. Our classifier is based on the fuzzy logic theory which can incorporate human knowledge into the process of inference. In this way, we are able to consult experts and create a system which partially mimics human reasoning. The power of the classifier will be demonstrated with two classes of real life underwater acoustic transients.
{"title":"A fuzzy logic based underwater acoustic transient classifier","authors":"T. Leung, Pr White","doi":"10.1109/DSPWS.1996.555570","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555570","url":null,"abstract":"This paper discusses a fuzzy logic based system developed to detect and classify a series of real life underwater acoustic transients. Signals received in the ocean are highly contaminated by various sources. While a trained sonar operator may classify transients without too difficulty, automatic classification by traditional template matching does not give satisfactory results. Our classifier is based on the fuzzy logic theory which can incorporate human knowledge into the process of inference. In this way, we are able to consult experts and create a system which partially mimics human reasoning. The power of the classifier will be demonstrated with two classes of real life underwater acoustic transients.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116945182","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555515
G. Shachor, M. Porat
Motivated by the human visual system, where both spectral phase and overlapping receptive fields play major roles, we develop conditions for unique representation of a signal by a combination of its spectral (Fourier) phase and spatial samples. Methods for signal reconstruction from the given combined information are introduced. It is concluded that most of the previous results in the area of Fourier phase-only representation and reconstruction are special cases of the general approach presented here.
{"title":"Signal representation and reconstruction from partial information in the position-frequency space","authors":"G. Shachor, M. Porat","doi":"10.1109/DSPWS.1996.555515","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555515","url":null,"abstract":"Motivated by the human visual system, where both spectral phase and overlapping receptive fields play major roles, we develop conditions for unique representation of a signal by a combination of its spectral (Fourier) phase and spatial samples. Methods for signal reconstruction from the given combined information are introduced. It is concluded that most of the previous results in the area of Fourier phase-only representation and reconstruction are special cases of the general approach presented here.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"113 51","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"120826587","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555540
J. Sorelius, R. Moses, T. Soderstrom
We consider the impact of bandwidth on narrowband direction-of-arrival (DOA) estimation using an array of sensors. We derive expressions for the DOA bias for three array processing algorithms: MUSIC, ESPRIT, and weighted subspace fitting. The bias expressions are found by a perturbation analysis of these algorithms for small relative bandwidths of the sources. We compare the perturbation-based bias predictions to actual bias for some cases of interest.
{"title":"Effects of nonzero bandwidth on direction of arrival estimators in array signal processing","authors":"J. Sorelius, R. Moses, T. Soderstrom","doi":"10.1109/DSPWS.1996.555540","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555540","url":null,"abstract":"We consider the impact of bandwidth on narrowband direction-of-arrival (DOA) estimation using an array of sensors. We derive expressions for the DOA bias for three array processing algorithms: MUSIC, ESPRIT, and weighted subspace fitting. The bias expressions are found by a perturbation analysis of these algorithms for small relative bandwidths of the sources. We compare the perturbation-based bias predictions to actual bias for some cases of interest.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"218 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122702436","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 : 1996-09-01DOI: 10.1109/DSPWS.1996.555488
H. Yasukawa
This paper describes a system that can enhance the quality of speech signals that have been severely band limited during transmission. We have already proposed a spectrum widening method that utilizes aliasing in sampling rate conversion with digital filtering for spectrum shaping. This paper proposes a new method that offers improved performance in terms of spectrum distortion characteristics. Implementation procedures are clarified, and its performance is discussed. The proposed method can effectively enhance speech quality.
{"title":"A simple method of broad band speech recovery from narrow band speech for quality enhancement","authors":"H. Yasukawa","doi":"10.1109/DSPWS.1996.555488","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555488","url":null,"abstract":"This paper describes a system that can enhance the quality of speech signals that have been severely band limited during transmission. We have already proposed a spectrum widening method that utilizes aliasing in sampling rate conversion with digital filtering for spectrum shaping. This paper proposes a new method that offers improved performance in terms of spectrum distortion characteristics. Implementation procedures are clarified, and its performance is discussed. The proposed method can effectively enhance speech quality.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"363 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122845081","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：