Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028157
L. Daudet, M. Sandler, B. Torrésani
We present in this paper the advantages of using overcomplete sets for the representation and coding of audio signals. More specifically, we have investigated hybrid signal models of the type {tonal+transients+stochastic residual} where these features can be simultaneously present at all times. The extraction of the two "deterministic" components (tonal and transient parts) is made in such a way that each of them can be represented in a compact way in appropriate orthogonal basis (MDCT and dyadic wavelets, respectively). The use of structured representations significantly reduces the potentially high cost of encoding of significance maps. Preliminary results show that this approach is well-adapted for transform coding of very high-quality sounds.
{"title":"Audio representations on overcomplete sets","authors":"L. Daudet, M. Sandler, B. Torrésani","doi":"10.1109/ICDSP.2002.1028157","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028157","url":null,"abstract":"We present in this paper the advantages of using overcomplete sets for the representation and coding of audio signals. More specifically, we have investigated hybrid signal models of the type {tonal+transients+stochastic residual} where these features can be simultaneously present at all times. The extraction of the two \"deterministic\" components (tonal and transient parts) is made in such a way that each of them can be represented in a compact way in appropriate orthogonal basis (MDCT and dyadic wavelets, respectively). The use of structured representations significantly reduces the potentially high cost of encoding of significance maps. Preliminary results show that this approach is well-adapted for transform coding of very high-quality sounds.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"48 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127098226","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028224
M. D. Galanis, A. Papazacharias, E. Zigouris
In this paper a course on real-time DSP systems design and implementation is presented which makes use of the MATLAB and the Texas Instruments TMS320C621 I DSP Starter Kit (DSK). The students attending this course get familiarized with the operation of DSP algorithms on an embedded digital signal processor. This course emphasizes the issue of the transition from an advanced design and simulation environment, like MATLAB, to a DSP software environment, like Code Composer Studio IDE.
本文介绍了一门利用MATLAB和德州仪器TMS320C621 I DSP Starter Kit (DSK)进行实时DSP系统设计与实现的课程。通过本课程的学习，学生将熟悉DSP算法在嵌入式数字信号处理器上的操作。本课程强调从高级设计和仿真环境(如MATLAB)到DSP软件环境(如Code Composer Studio IDE)的过渡问题。
{"title":"A DSP course for real-time systems design and implementation based on the TMS320C6211 DSK","authors":"M. D. Galanis, A. Papazacharias, E. Zigouris","doi":"10.1109/ICDSP.2002.1028224","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028224","url":null,"abstract":"In this paper a course on real-time DSP systems design and implementation is presented which makes use of the MATLAB and the Texas Instruments TMS320C621 I DSP Starter Kit (DSK). The students attending this course get familiarized with the operation of DSP algorithms on an embedded digital signal processor. This course emphasizes the issue of the transition from an advanced design and simulation environment, like MATLAB, to a DSP software environment, like Code Composer Studio IDE.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124983540","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028241
B. Smolka, M. Szczepański
A novel approach to the problem of edge preserving smoothing, which allows an image to be broken into a set of homogeneous regions, is proposed and evaluated. The new algorithm is based on combined forward and backward anisotropic diffusion with an incorporated time dependent cooling process. This method is able to remove image noise efficiently while preserving and enhancing image edges.
{"title":"Forward and backward anisotropic diffusion filtering for color image enhancement","authors":"B. Smolka, M. Szczepański","doi":"10.1109/ICDSP.2002.1028241","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028241","url":null,"abstract":"A novel approach to the problem of edge preserving smoothing, which allows an image to be broken into a set of homogeneous regions, is proposed and evaluated. The new algorithm is based on combined forward and backward anisotropic diffusion with an incorporated time dependent cooling process. This method is able to remove image noise efficiently while preserving and enhancing image edges.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124996159","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028210
A. Tarczynski
Nonuniform sampling of signals, if properly used, may allow performing digital signal processing that does not suffer from the undesired effects of aliasing. This paper presents the results of research on estimating spectra of signals whose samples were taken at randomly distributed sampling instants. The paper determines the conditions under which a DFT-like spectrum estimator provides an unbiased approximation of the spectrum of the original continuous-time signal in an unlimited range of frequencies. It also provides analysis of the accuracy of such spectral estimation and identifies factors which decide the quality of the results. Not every random sampling scheme is suitable for performing alias free signal processing. The analysis presented here shows how to generate two suitable sampling schemes. Numerical examples illustrate the main thesis of the paper.
{"title":"Spectrum estimation of nonuniformly sampled signals","authors":"A. Tarczynski","doi":"10.1109/ICDSP.2002.1028210","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028210","url":null,"abstract":"Nonuniform sampling of signals, if properly used, may allow performing digital signal processing that does not suffer from the undesired effects of aliasing. This paper presents the results of research on estimating spectra of signals whose samples were taken at randomly distributed sampling instants. The paper determines the conditions under which a DFT-like spectrum estimator provides an unbiased approximation of the spectrum of the original continuous-time signal in an unlimited range of frequencies. It also provides analysis of the accuracy of such spectral estimation and identifies factors which decide the quality of the results. Not every random sampling scheme is suitable for performing alias free signal processing. The analysis presented here shows how to generate two suitable sampling schemes. Numerical examples illustrate the main thesis of the paper.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"604 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123235919","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028251
lsmo Kauppinen
Computationally efficient methods for detecting non-Gaussian impulsive noise in digital speech and audio signals are presented. The aim of the detection is to find the errors without false detections in the case of e.g. percussive sounds in music signal or stop-consonants in speech signal. Various methods for computing a detection signal and a threshold curve are studied and tested. The detection can be applied in real time to a digital data stream.
{"title":"Methods for detecting impulsive noise in speech and audio signals","authors":"lsmo Kauppinen","doi":"10.1109/ICDSP.2002.1028251","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028251","url":null,"abstract":"Computationally efficient methods for detecting non-Gaussian impulsive noise in digital speech and audio signals are presented. The aim of the detection is to find the errors without false detections in the case of e.g. percussive sounds in music signal or stop-consonants in speech signal. Various methods for computing a detection signal and a threshold curve are studied and tested. The detection can be applied in real time to a digital data stream.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"56 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123266809","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028178
S. Chan, Yue-Xian Zou
We present the convergence analysis of the recursive least M-estimate (RLM) adaptive filter algorithm, which was recently proposed for robust adaptive filtering in the impulse noise environment. The mean and mean squares behaviors of the RLM algorithm, based on the modified Huber M-estimate function (MHF), in the contaminated Gaussian (CG) noise model are analyzed. Close-form expressions are derived. The simulation and theoretical results agree very well with each other and suggest that the RLM algorithm is more robust than the RLS algorithm under the CG noise model.
{"title":"Convergence analysis of the recursive least M-estimate adaptive filtering algorithm for impulse noise suppression","authors":"S. Chan, Yue-Xian Zou","doi":"10.1109/ICDSP.2002.1028178","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028178","url":null,"abstract":"We present the convergence analysis of the recursive least M-estimate (RLM) adaptive filter algorithm, which was recently proposed for robust adaptive filtering in the impulse noise environment. The mean and mean squares behaviors of the RLM algorithm, based on the modified Huber M-estimate function (MHF), in the contaminated Gaussian (CG) noise model are analyzed. Close-form expressions are derived. The simulation and theoretical results agree very well with each other and suggest that the RLM algorithm is more robust than the RLS algorithm under the CG noise model.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"21 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125251513","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028257
F. Beritelli, S. Casale, S. Serrano
The paper deals with a simple speaker-dependent (SD) isolated word recognition (IWR) system based on template-based pattern matching. Different algorithms for storing and calculating the distortion between models and examples of words to be recognised are analysed. More specifically, the paper proposes a new algorithm that enhances performance with a slight increase in computational load and the amount of memory needed to store the models as compared with a traditional VQ-based approach. The results obtained in tests are given in terms of recognition rate, using the TIMIT-46 database with various type of background noise and different SNRs.
{"title":"A robust speaker dependent algorithm for isolated word recognition","authors":"F. Beritelli, S. Casale, S. Serrano","doi":"10.1109/ICDSP.2002.1028257","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028257","url":null,"abstract":"The paper deals with a simple speaker-dependent (SD) isolated word recognition (IWR) system based on template-based pattern matching. Different algorithms for storing and calculating the distortion between models and examples of words to be recognised are analysed. More specifically, the paper proposes a new algorithm that enhances performance with a slight increase in computational load and the amount of memory needed to store the models as compared with a traditional VQ-based approach. The results obtained in tests are given in terms of recognition rate, using the TIMIT-46 database with various type of background noise and different SNRs.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"56 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126761461","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028344
Joni-Kristian Kämäräinen, V. Kyrki, H. Kälviäinen
The choice of features for invariant object recognition is one of the most essential problems in computer vision. The authors have previously proposed Gabor (1946) filtering based feature extraction methods which have been successfully applied in invariant object recognition. In this study, the Gabor filtering based feature extraction is further analysed in terms of distortion tolerance which is an essential property for many applications. Experiments indicate that an accurate recognition can be achieved in the presence of significant amounts of distortions.
{"title":"Noise tolerant object recognition using Gabor filtering","authors":"Joni-Kristian Kämäräinen, V. Kyrki, H. Kälviäinen","doi":"10.1109/ICDSP.2002.1028344","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028344","url":null,"abstract":"The choice of features for invariant object recognition is one of the most essential problems in computer vision. The authors have previously proposed Gabor (1946) filtering based feature extraction methods which have been successfully applied in invariant object recognition. In this study, the Gabor filtering based feature extraction is further analysed in terms of distortion tolerance which is an essential property for many applications. Experiments indicate that an accurate recognition can be achieved in the presence of significant amounts of distortions.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"17 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122407988","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028268
T. Fliess, H. Jentschel
Beside measurement devices and noise generators transmission lines or devices modeling the electrical behavior of these lines are required within test environments for xDSL equipment. State-of-the-art line simulators are based on passive networks or on active analog circuitry. We propose a line simulation concept applying real time digital signal processing techniques. This new approach provides a high flexibility and simulation accuracy.
{"title":"Digital real time line simulator","authors":"T. Fliess, H. Jentschel","doi":"10.1109/ICDSP.2002.1028268","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028268","url":null,"abstract":"Beside measurement devices and noise generators transmission lines or devices modeling the electrical behavior of these lines are required within test environments for xDSL equipment. State-of-the-art line simulators are based on passive networks or on active analog circuitry. We propose a line simulation concept applying real time digital signal processing techniques. This new approach provides a high flexibility and simulation accuracy.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"45 1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114456960","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 : 2002-11-07DOI: 10.1109/ICDSP.2002.1028185
K. Ntalianis, A. Doulamis, N. Doulamis, S. Kollias
Two unsupervised video object segmentation techniques are proposed and are compared in terms of computational cost and segmentation quality. Both methods are based on the exploitation of depth information. In particular a depth segments map is initially estimated by analyzing a stereoscopic pair of frames and applying a segmentation algorithm. Next, considering the first "constrained fusion of color segments" (CFCS) approach, color segmentation is performed to one of the stereo pairs and video objects are extracted by fusing color segments according to depth similarity. In the second method an active contour is automatically initialized onto the boundary of each depth segment, according to a fitness function that considers different color areas and preserves the shapes of depth segments' boundaries. Then the active contour moves onto a grid to extract the video object. Experiments on real stereoscopic sequences exhibit the speed and accuracy of the proposed schemes.
{"title":"Unsupervised segmentation of stereoscopic video objects: investigation of two depth-based approaches","authors":"K. Ntalianis, A. Doulamis, N. Doulamis, S. Kollias","doi":"10.1109/ICDSP.2002.1028185","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028185","url":null,"abstract":"Two unsupervised video object segmentation techniques are proposed and are compared in terms of computational cost and segmentation quality. Both methods are based on the exploitation of depth information. In particular a depth segments map is initially estimated by analyzing a stereoscopic pair of frames and applying a segmentation algorithm. Next, considering the first \"constrained fusion of color segments\" (CFCS) approach, color segmentation is performed to one of the stereo pairs and video objects are extracted by fusing color segments according to depth similarity. In the second method an active contour is automatically initialized onto the boundary of each depth segment, according to a fitness function that considers different color areas and preserves the shapes of depth segments' boundaries. Then the active contour moves onto a grid to extract the video object. Experiments on real stereoscopic sequences exhibit the speed and accuracy of the proposed schemes.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"56 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122165062","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：