Pub Date : 2000-10-29DOI: 10.1109/ACSSC.2000.910646
M.S. Patichis, H. Petropoulos, W. Brooks
MRI brain images are characterized by non-stationary components that make fully automated segmentation a challenging task. An AM-FM model is used to model these non-stationarities. Using the AM-FM model, a new, fully automated texture segmentation system is used to automatically segment the cerebellum from a 3-D set of MRI brain images.
{"title":"MRI brain image segmentation using an AM-FM model","authors":"M.S. Patichis, H. Petropoulos, W. Brooks","doi":"10.1109/ACSSC.2000.910646","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.910646","url":null,"abstract":"MRI brain images are characterized by non-stationary components that make fully automated segmentation a challenging task. An AM-FM model is used to model these non-stationarities. Using the AM-FM model, a new, fully automated texture segmentation system is used to automatically segment the cerebellum from a 3-D set of MRI brain images.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"6 1","pages":"906-910 vol.2"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"89067992","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.910758
R. Prakash, V. Veeravalli
A reverse link random access system is considered where CDMA with random spreading is used for reset separation. The receiver consists of either a matched filter (MF) or a minimum mean squared error (MMSE) detector followed by autonomous forward error correction (FEC) decoders for each user. The random access strategy combines slotted ALOHA with incremental redundancy (IR). Such a system is defined as a code division random multiple access (CDRMA) system. Two types of IR, namely code combining and maximal ratio combining (MRC) are considered. Bounds on the throughput of a CDRMA system are obtained for different detectors and IR schemes, when the number of users K and the spreading factor N, are both large (K,N/spl rarr//spl infin/, K/N=/spl alpha/). These bounds are derived using known results on the information theoretic capacity for a user within a slot. The bound on the throughput of a CDRMA system is shown to be equal to the bound on the throughput of an equivalent fixed access (conventional) CDMA system.
{"title":"Analysis of code division random multiple access systems with packet combining","authors":"R. Prakash, V. Veeravalli","doi":"10.1109/ACSSC.2000.910758","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.910758","url":null,"abstract":"A reverse link random access system is considered where CDMA with random spreading is used for reset separation. The receiver consists of either a matched filter (MF) or a minimum mean squared error (MMSE) detector followed by autonomous forward error correction (FEC) decoders for each user. The random access strategy combines slotted ALOHA with incremental redundancy (IR). Such a system is defined as a code division random multiple access (CDRMA) system. Two types of IR, namely code combining and maximal ratio combining (MRC) are considered. Bounds on the throughput of a CDRMA system are obtained for different detectors and IR schemes, when the number of users K and the spreading factor N, are both large (K,N/spl rarr//spl infin/, K/N=/spl alpha/). These bounds are derived using known results on the information theoretic capacity for a user within a slot. The bound on the throughput of a CDRMA system is shown to be equal to the bound on the throughput of an equivalent fixed access (conventional) CDMA system.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"25 1","pages":"1225-1229 vol.2"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"90497974","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.910991
N. Magotra, S. Kamath, F. Livingston, M. Ho
This paper describes the development and real-time, fixed-point implementation of a Multiband Dynamic Range Compression (MDRC) algorithm on a 16 bit fixed-point platform-Texas Instrument's TMS320C54X. This algorithm has been designed primarily for use in hearing aid applications. It has been implemented on a prototype evaluation system that operates by default in a stereo mode with a minimum sampling rate of 20 KHz per channel. The paper also describes a new algorithm development software environment developed at Texas Instruments to facilitate the task of implementing complex Digital Signal Processing (DSP) algorithms on its DSP chips. This tool is known as eXpress DSP Algorithm Standard (XDAIS). Developing an algorithm that is XDAIS compliant facilitates the easy integration of the algorithm into a system. It also simplifies porting the algorithm to other applications or systems. For example, the XDAIS compliant MDRC algorithm, designed originally for hearing aid applications, could easily be ported for other audio or sensor data processing applications.
{"title":"Development and fixed-point implementation of a multiband dynamic range compression (MDRC) algorithm","authors":"N. Magotra, S. Kamath, F. Livingston, M. Ho","doi":"10.1109/ACSSC.2000.910991","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.910991","url":null,"abstract":"This paper describes the development and real-time, fixed-point implementation of a Multiband Dynamic Range Compression (MDRC) algorithm on a 16 bit fixed-point platform-Texas Instrument's TMS320C54X. This algorithm has been designed primarily for use in hearing aid applications. It has been implemented on a prototype evaluation system that operates by default in a stereo mode with a minimum sampling rate of 20 KHz per channel. The paper also describes a new algorithm development software environment developed at Texas Instruments to facilitate the task of implementing complex Digital Signal Processing (DSP) algorithms on its DSP chips. This tool is known as eXpress DSP Algorithm Standard (XDAIS). Developing an algorithm that is XDAIS compliant facilitates the easy integration of the algorithm into a system. It also simplifies porting the algorithm to other applications or systems. For example, the XDAIS compliant MDRC algorithm, designed originally for hearing aid applications, could easily be ported for other audio or sensor data processing applications.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"240 ","pages":"428-432 vol.1"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"91450716","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.910633
T. Biedka, J. Reed, W. Tranter
A spatial signature is used to model a communication signal that is received by an array of antennas in a multipath environment. If the transmitted waveform is not known, a blind estimate of the unknown spatial signature may be obtained by cross-correlating an estimate of the transmitted waveform with the received data. This paper presents the mean and variance of a blind spatial signature estimator that is obtained by either a constant modulus mapping or by mapping onto a known finite alphabet. These results are compared to those for an estimator that exploits a known waveform.
{"title":"Statistics of blind spatial signature estimators","authors":"T. Biedka, J. Reed, W. Tranter","doi":"10.1109/ACSSC.2000.910633","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.910633","url":null,"abstract":"A spatial signature is used to model a communication signal that is received by an array of antennas in a multipath environment. If the transmitted waveform is not known, a blind estimate of the unknown spatial signature may be obtained by cross-correlating an estimate of the transmitted waveform with the received data. This paper presents the mean and variance of a blind spatial signature estimator that is obtained by either a constant modulus mapping or by mapping onto a known finite alphabet. These results are compared to those for an estimator that exploits a known waveform.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"93 1","pages":"847-850 vol.2"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"83163266","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.910919
Joseph, S. Acton
Segmentation-based image and video coding is desirable for many multimedia applications due to the additional functionality provided by object-based representation. Methods of object-based coding have generally treated the segmentation and encoding processes as separate problems. Here, we present an integrated segmentation and coding method unified by the theoretical structure of morphological local monotonicity. This unified segmentation/coding scheme utilizes morphological operators within a nonlinear scale-space to generate a segmentation. The segmented regions are independently coded and reconstructed using morphological generalization of Laplace's equation in a multiresolution framework. The coding procedure is appropriate for non-textured imagery and avoids arbitrarily chosen constants. Examples are given for two-dimensional grayscale imagery.
{"title":"Segmentation-based image coding by morphological local monotonicity","authors":"Joseph, S. Acton","doi":"10.1109/ACSSC.2000.910919","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.910919","url":null,"abstract":"Segmentation-based image and video coding is desirable for many multimedia applications due to the additional functionality provided by object-based representation. Methods of object-based coding have generally treated the segmentation and encoding processes as separate problems. Here, we present an integrated segmentation and coding method unified by the theoretical structure of morphological local monotonicity. This unified segmentation/coding scheme utilizes morphological operators within a nonlinear scale-space to generate a segmentation. The segmented regions are independently coded and reconstructed using morphological generalization of Laplace's equation in a multiresolution framework. The coding procedure is appropriate for non-textured imagery and avoids arbitrarily chosen constants. Examples are given for two-dimensional grayscale imagery.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"33 1","pages":"65-69 vol.1"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"81208889","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.911232
W. Bastiaan Kleijn, A. Jefremov, M. Murthi
Speech is well described by a source-filter model. The source properties are critical for good quality reconstructed speech. We describe a source model which facilitates both low-rate coding and signal modification. The source signal is described by means of pitch-synchronous frame expansions, with different subsets of the coefficients corresponding to so-called voiced and unvoiced components. To obtain a perceptually plausible voiced-unvoiced decomposition even at speech onsets, our frame functions adapt to the signal. The generation of the unvoiced component consists of the replacement of the corresponding coefficients with realizations of a random variable with similar statistics. Existing sinusoidal and waveform-interpolation excitation models form approximations to the presented procedure.
{"title":"Identification and reconstruction of the unvoiced component in speech","authors":"W. Bastiaan Kleijn, A. Jefremov, M. Murthi","doi":"10.1109/ACSSC.2000.911232","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.911232","url":null,"abstract":"Speech is well described by a source-filter model. The source properties are critical for good quality reconstructed speech. We describe a source model which facilitates both low-rate coding and signal modification. The source signal is described by means of pitch-synchronous frame expansions, with different subsets of the coefficients corresponding to so-called voiced and unvoiced components. To obtain a perceptually plausible voiced-unvoiced decomposition even at speech onsets, our frame functions adapt to the signal. The generation of the unvoiced component consists of the replacement of the corresponding coefficients with realizations of a random variable with similar statistics. Existing sinusoidal and waveform-interpolation excitation models form approximations to the presented procedure.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"4 1","pages":"1459-1463 vol.2"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"88759769","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.910690
W. Melvin, M. J. Callahan, M. Wicks
The detection performance of airborne bistatic radar depends on suitable clutter mitigation techniques. Space-time adaptive processing (STAP) is a candidate for suppressing bistatic clutter. Since STAP implementation relies on secondary data for covariance estimation, the nonstationary nature of typical bistatic angle-Doppler contours over range is a point of concern. In this paper we find that bistatic STAP suffers greater performance degradation than commonly observed in monostatic STAP analyses. Additionally we briefly examine approaches to enhance bistatic STAP capability.
{"title":"Adaptive clutter cancellation in bistatic radar","authors":"W. Melvin, M. J. Callahan, M. Wicks","doi":"10.1109/ACSSC.2000.910690","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.910690","url":null,"abstract":"The detection performance of airborne bistatic radar depends on suitable clutter mitigation techniques. Space-time adaptive processing (STAP) is a candidate for suppressing bistatic clutter. Since STAP implementation relies on secondary data for covariance estimation, the nonstationary nature of typical bistatic angle-Doppler contours over range is a point of concern. In this paper we find that bistatic STAP suffers greater performance degradation than commonly observed in monostatic STAP analyses. Additionally we briefly examine approaches to enhance bistatic STAP capability.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"61 1","pages":"1125-1130 vol.2"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"84808278","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.911266
C. Lindquist, C. Corral
Filter nomographs have proven to be effective design tools in determining filter order and evaluating design tradeoffs. This paper presents nomographs for filters that are maximally flat magnitude beyond the origin (MFMBO). It is shown that MFMBO filters display transitional characteristics and can be designed for different shaping factors. The proposed nomographs con be used to determine filter order and degrees of freedom in a MFMBO filter design.
{"title":"Nomographs for MFMBO filters (maximally flat magnitude beyond the origin)","authors":"C. Lindquist, C. Corral","doi":"10.1109/ACSSC.2000.911266","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.911266","url":null,"abstract":"Filter nomographs have proven to be effective design tools in determining filter order and evaluating design tradeoffs. This paper presents nomographs for filters that are maximally flat magnitude beyond the origin (MFMBO). It is shown that MFMBO filters display transitional characteristics and can be designed for different shaping factors. The proposed nomographs con be used to determine filter order and degrees of freedom in a MFMBO filter design.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"1 1","pages":"1635-1638 vol.2"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"86701807","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.911274
A. Faroqui, V.G. Okobdzija
A programmable data-path that supports MPEG standards Synthetic & Natural Hybrid video Coding (SNHC) is presented. It can support a maximum of 16 parallel SIMD integer operations and 2 parallel SIMD floating-point operations. Two new instructions were added in order to increase the execution of 3D graphics and SNHC as well as to speed up IDCT, FFT, and other media signal processing algorithms. These operations are implemented by re-using the hardware without significant increase in area and delay. The datapath has been modeled in Verilog using 0.25u CMOS library and synthesized using Synopsys. All operations are single-cycle running at 200 MHz.
{"title":"A programmable data-path for MPEG-4 and natural hybrid video coding","authors":"A. Faroqui, V.G. Okobdzija","doi":"10.1109/ACSSC.2000.911274","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.911274","url":null,"abstract":"A programmable data-path that supports MPEG standards Synthetic & Natural Hybrid video Coding (SNHC) is presented. It can support a maximum of 16 parallel SIMD integer operations and 2 parallel SIMD floating-point operations. Two new instructions were added in order to increase the execution of 3D graphics and SNHC as well as to speed up IDCT, FFT, and other media signal processing algorithms. These operations are implemented by re-using the hardware without significant increase in area and delay. The datapath has been modeled in Verilog using 0.25u CMOS library and synthesized using Synopsys. All operations are single-cycle running at 200 MHz.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"49 1","pages":"1675-1678 vol.2"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"89872795","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 : 2000-10-29DOI: 10.1109/ACSSC.2000.910951
A. E. Brito, S. Cabrera
The adaptive weighted norm extrapolation (AWNE) method is well suited for recovering sinusoidal signals from short data records since they have well concentrated spectral representations. This paper analyzes the performance of the algorithm in the recovery of sinusoidal signals; validates the assumption that the result of the algorithm is a windowed version of the original; and derives a form for this imposed window. The role of the autocorrelation of the window used in the AWNE algorithm plays a key role in our analysis. We also provide a new interpretation of the AWNE method as a best subspace selection method for signal approximation.
{"title":"Analysis of an adaptive extrapolation algorithm on the recovery of harmonic signals","authors":"A. E. Brito, S. Cabrera","doi":"10.1109/ACSSC.2000.910951","DOIUrl":"https://doi.org/10.1109/ACSSC.2000.910951","url":null,"abstract":"The adaptive weighted norm extrapolation (AWNE) method is well suited for recovering sinusoidal signals from short data records since they have well concentrated spectral representations. This paper analyzes the performance of the algorithm in the recovery of sinusoidal signals; validates the assumption that the result of the algorithm is a windowed version of the original; and derives a form for this imposed window. The role of the autocorrelation of the window used in the AWNE algorithm plays a key role in our analysis. We also provide a new interpretation of the AWNE method as a best subspace selection method for signal approximation.","PeriodicalId":10581,"journal":{"name":"Conference Record of the Thirty-Fourth Asilomar Conference on Signals, Systems and Computers (Cat. No.00CH37154)","volume":"15 1","pages":"233-237 vol.1"},"PeriodicalIF":0.0,"publicationDate":"2000-10-29","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"89908757","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}
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