Pub Date : 2008-05-12DOI: 10.1109/ICASSP.2008.4517668
V. Sanchez, P. Nasiopoulos, R. Abugharbieh
Dynamic volumetric (four dimensional- 4D) medical images are typically huge in file size and require a vast amount of resources for storage and transmission purposes. In this paper, we propose an efficient lossless compression method for 4D medical images that is based on a multi-frame motion compensation process employing a 4D search, variable block- sizes and bi-directional prediction. Data redundancies are reduced by recursively applying multi-frame motion compensation in the spatial and temporal dimensions. The proposed method also uses a novel differential coding algorithm to reduce redundancies in motion vectors and a new context-based adaptive binary arithmetic coder (CABAC) for compression of the residual data. Performance evaluations on real medical images of varying modality resulted in lossless compression ratios of up to 16:1.
{"title":"Efficient 4D motion compensated lossless compression of dynamic volumetric medical image data","authors":"V. Sanchez, P. Nasiopoulos, R. Abugharbieh","doi":"10.1109/ICASSP.2008.4517668","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517668","url":null,"abstract":"Dynamic volumetric (four dimensional- 4D) medical images are typically huge in file size and require a vast amount of resources for storage and transmission purposes. In this paper, we propose an efficient lossless compression method for 4D medical images that is based on a multi-frame motion compensation process employing a 4D search, variable block- sizes and bi-directional prediction. Data redundancies are reduced by recursively applying multi-frame motion compensation in the spatial and temporal dimensions. The proposed method also uses a novel differential coding algorithm to reduce redundancies in motion vectors and a new context-based adaptive binary arithmetic coder (CABAC) for compression of the residual data. Performance evaluations on real medical images of varying modality resulted in lossless compression ratios of up to 16:1.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116087820","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4518866
Joseph Meehan
A dedicated media processor is used in many camera phones to accelerate video and image processing. Increased demand for higher pixel resolution and higher quality image and video processing necessitates dramatically increased signal processing capability. To provide the increased performance at acceptable cost and power requires redesign of the traditional architecture. By wisely partitioning algorithms across programmable and fixed- function blocks, the performance goals can be met while still maintaining flexibility for new feature requirements and new standards. In this paper we provide an overview of media processor architectures for camera phones and describe the system architecture, power, and performance. We also address the challenges in supporting new imaging trends and high resolution video at low power and cost.
{"title":"Media processor architecture for video and imaging on camera phones","authors":"Joseph Meehan","doi":"10.1109/ICASSP.2008.4518866","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4518866","url":null,"abstract":"A dedicated media processor is used in many camera phones to accelerate video and image processing. Increased demand for higher pixel resolution and higher quality image and video processing necessitates dramatically increased signal processing capability. To provide the increased performance at acceptable cost and power requires redesign of the traditional architecture. By wisely partitioning algorithms across programmable and fixed- function blocks, the performance goals can be met while still maintaining flexibility for new feature requirements and new standards. In this paper we provide an overview of media processor architectures for camera phones and describe the system architecture, power, and performance. We also address the challenges in supporting new imaging trends and high resolution video at low power and cost.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116575261","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517565
M. Schlosser
An acoustic detector for film slates is proposed to assist a human operator with the synchronization of audio and video in post-production. To be computationally efficient, the signal analysis is restricted to time-domain features. Although the features are statistically dependent, separate classifiers are trained for each of them. The statistical dependence is taken into account during the combination of the log-likelihood ratios provided by the individual classifiers. The overall confidence in a classification is determined as a weighted sum of the individual log-likelihood ratios, where the weights depend on the correlation between the different features. Experimental results for real-world recordings from film sets show that the confidence measures allow for a fast identification of the film slates while minimizing the interference from false detections.
{"title":"Confidence measures for acoustic detection of film slates based on time-domain features","authors":"M. Schlosser","doi":"10.1109/ICASSP.2008.4517565","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517565","url":null,"abstract":"An acoustic detector for film slates is proposed to assist a human operator with the synchronization of audio and video in post-production. To be computationally efficient, the signal analysis is restricted to time-domain features. Although the features are statistically dependent, separate classifiers are trained for each of them. The statistical dependence is taken into account during the combination of the log-likelihood ratios provided by the individual classifiers. The overall confidence in a classification is determined as a weighted sum of the individual log-likelihood ratios, where the weights depend on the correlation between the different features. Experimental results for real-world recordings from film sets show that the confidence measures allow for a fast identification of the film slates while minimizing the interference from false detections.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116583615","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517887
Sheng-Yu Peng, Yu-Chi Tsao, P. Hasler, David V. Anderson
A 16 x 16 programmable analog radial-basis-function (RBF) based classifier is demonstrated. The distribution of each feature is modeled by a Gaussian function, which is realized by a proposed floating-gate bump circuit having bell-shaped transfer characteristics. The maximum likelihood, mean, and variance of the distribution are stored in floating-gate transistors and are independently programmable. By cascading these floating-gate bump circuits, the overall transfer characteristics approximate a multivariate Gaussian distribution with a diagonal covariance matrix. An array of these circuits constitutes a compact RBF-based classifier. When followed by a winner-take-all circuit, the analog classifier can implement vector quantization. Automatic gender identification is implemented on a 16 x 16 analog vector quantizer chip as one possible audio application of this work. The performance of the analog classifier is comparable to that of digital counter -parts. The proposed approach can be at least two orders of magnitude more power efficient than the digital microprocessors at the same task.
{"title":"A programmable analog radial-basis-function based classifier","authors":"Sheng-Yu Peng, Yu-Chi Tsao, P. Hasler, David V. Anderson","doi":"10.1109/ICASSP.2008.4517887","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517887","url":null,"abstract":"A 16 x 16 programmable analog radial-basis-function (RBF) based classifier is demonstrated. The distribution of each feature is modeled by a Gaussian function, which is realized by a proposed floating-gate bump circuit having bell-shaped transfer characteristics. The maximum likelihood, mean, and variance of the distribution are stored in floating-gate transistors and are independently programmable. By cascading these floating-gate bump circuits, the overall transfer characteristics approximate a multivariate Gaussian distribution with a diagonal covariance matrix. An array of these circuits constitutes a compact RBF-based classifier. When followed by a winner-take-all circuit, the analog classifier can implement vector quantization. Automatic gender identification is implemented on a 16 x 16 analog vector quantizer chip as one possible audio application of this work. The performance of the analog classifier is comparable to that of digital counter -parts. The proposed approach can be at least two orders of magnitude more power efficient than the digital microprocessors at the same task.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122345941","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517844
Yu He, Kim-Hui Yap, Li Chen, Lap-Pui Chau
This paper proposes a new regularization scheme to address blind color image deconvolution. Conventional blind monochromatic image deconvolution algorithms handle each color channel independently, thereby ignoring the inter-channel correlation present in the color images. Further, most existing blind color deconvolution algorithms do not take the parametric information of the blurs into consideration. In view of these, a regularization scheme is proposed to perform blind color image deconvolution. A new regularization operator is developed in the blur domain. A reinforcement blur modeling scheme is adopted to evaluate the relevance of manifold parametric blur structures, and the information is integrated into the deconvolution scheme. In addition, a regularization scheme for image is developed to recover edges of color images and reduce color artifacts. Experimental results show that the method is able to achieve satisfactory restored color images under noisy environment.
{"title":"A new color image regularization scheme for blind image deconvolution","authors":"Yu He, Kim-Hui Yap, Li Chen, Lap-Pui Chau","doi":"10.1109/ICASSP.2008.4517844","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517844","url":null,"abstract":"This paper proposes a new regularization scheme to address blind color image deconvolution. Conventional blind monochromatic image deconvolution algorithms handle each color channel independently, thereby ignoring the inter-channel correlation present in the color images. Further, most existing blind color deconvolution algorithms do not take the parametric information of the blurs into consideration. In view of these, a regularization scheme is proposed to perform blind color image deconvolution. A new regularization operator is developed in the blur domain. A reinforcement blur modeling scheme is adopted to evaluate the relevance of manifold parametric blur structures, and the information is integrated into the deconvolution scheme. In addition, a regularization scheme for image is developed to recover edges of color images and reduce color artifacts. Experimental results show that the method is able to achieve satisfactory restored color images under noisy environment.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122637909","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517617
Flavio P. Ribeiro, V. Nascimento
We present a computationally efficient method for sinusoidal noise cancellation based on the FXLMS algorithm. It features subsampling in order to increase convergence speed and decrease computational requirements, and most importantly, does not require extra noise added to the filter output for secondary path identification. In addition, it is robust to secondary path variations and in low SNR scenarios, which are frequently found in practical active noise control systems, features fast tracking and can be directly generalized to multichannel systems. We illustrate its operation with simulations.
{"title":"A robust and computationally efficient method for tonal active noise control using a simplified secondary path model","authors":"Flavio P. Ribeiro, V. Nascimento","doi":"10.1109/ICASSP.2008.4517617","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517617","url":null,"abstract":"We present a computationally efficient method for sinusoidal noise cancellation based on the FXLMS algorithm. It features subsampling in order to increase convergence speed and decrease computational requirements, and most importantly, does not require extra noise added to the filter output for secondary path identification. In addition, it is robust to secondary path variations and in low SNR scenarios, which are frequently found in practical active noise control systems, features fast tracking and can be directly generalized to multichannel systems. We illustrate its operation with simulations.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122817284","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517974
You-Seok Lee, YongJe Choi, Dong‐Guk Han, Ho-Yeon Kim, Hyoung-Nam Kim
Differential power analysis (DPA) has been known as an efficient attack for finding secret keys of cryptosystems but its efficiency may be lowered due to the misalignment of the acquired signals. Though the misalignment problem has been now solvable by various successful approaches in DPA, a lot of power traces are still required to find correct keys. Since the required number of power traces is directly connected with the efficiency of SCAs, we propose a key-search method even with relatively reduced number of power traces based on recognizing special patterns of the signal caused by cryptographic operations. Experimental results show that the proposed method is able to search correct keys with much smaller number of traces than the minimum number of traces with which the conventional methods of the energy-based DPA and frequency-based DPA succeed in finding keys.
{"title":"A nobel key-search method for side channel attacks based on pattern recognition","authors":"You-Seok Lee, YongJe Choi, Dong‐Guk Han, Ho-Yeon Kim, Hyoung-Nam Kim","doi":"10.1109/ICASSP.2008.4517974","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517974","url":null,"abstract":"Differential power analysis (DPA) has been known as an efficient attack for finding secret keys of cryptosystems but its efficiency may be lowered due to the misalignment of the acquired signals. Though the misalignment problem has been now solvable by various successful approaches in DPA, a lot of power traces are still required to find correct keys. Since the required number of power traces is directly connected with the efficiency of SCAs, we propose a key-search method even with relatively reduced number of power traces based on recognizing special patterns of the signal caused by cryptographic operations. Experimental results show that the proposed method is able to search correct keys with much smaller number of traces than the minimum number of traces with which the conventional methods of the energy-based DPA and frequency-based DPA succeed in finding keys.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114141380","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517990
Yi-Ou Li, Wei Wang, T. Adalı, V. Calhoun
In this work, we propose a scheme for joint blind source separation (BSS) of multiple datasets using canonical correlation analysis (CCA). The proposed scheme jointly extracts sources from each dataset in the order of between-set source correlations. We show that, when sources are uncorrelated within each dataset and correlated across different datasets only on corresponding indices, (i) CCA on two datasets achieves BSS when the sources from the two datasets have distinct between-set correlation coefficients, and (ii) CCA on multiple datasets (M-CCA) achieves BSS with a more relaxed condition on the between-set source correlation coefficients compared to CCA on two datasets. We present simulation results to demonstrate the properties of CCA and M-CCA on joint BSS. We apply M-CCA to group functional magnetic resonance imaging (fMRI) data acquired from several subjects performing a visuomotor task and obtain interesting brain activations as well as their correlation profiles across different subjects in the group.
{"title":"CCA for joint blind source separation of multiple datasets with application to group FMRI analysis","authors":"Yi-Ou Li, Wei Wang, T. Adalı, V. Calhoun","doi":"10.1109/ICASSP.2008.4517990","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517990","url":null,"abstract":"In this work, we propose a scheme for joint blind source separation (BSS) of multiple datasets using canonical correlation analysis (CCA). The proposed scheme jointly extracts sources from each dataset in the order of between-set source correlations. We show that, when sources are uncorrelated within each dataset and correlated across different datasets only on corresponding indices, (i) CCA on two datasets achieves BSS when the sources from the two datasets have distinct between-set correlation coefficients, and (ii) CCA on multiple datasets (M-CCA) achieves BSS with a more relaxed condition on the between-set source correlation coefficients compared to CCA on two datasets. We present simulation results to demonstrate the properties of CCA and M-CCA on joint BSS. We apply M-CCA to group functional magnetic resonance imaging (fMRI) data acquired from several subjects performing a visuomotor task and obtain interesting brain activations as well as their correlation profiles across different subjects in the group.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114251288","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4518610
Štefan Petrík, F. Pernkopf
Automatic phonetic reconstruction of medical dictations from non- literal and automatically recognized speech transcripts leads to closer-to-literal transcripts for training. In this paper, we introduce an extended alignment method assessing multiple levels of text segmentation and show how open issues like wrong segmentation in the recognized transcript can be resolved. Furthermore, the effect of context-dependent reconstruction and the phonetic similarity threshold on the quality of the reconstructed transcription is measured. Experiments show an increase in precision between 0.7% and 4.7% absolute without loss in recall for the combined system incorporating all of these techniques in comparison to the system in the previous work.
{"title":"Automatic phonetics-driven reconstruction of medical dictations on multiple levels of segmentation","authors":"Štefan Petrík, F. Pernkopf","doi":"10.1109/ICASSP.2008.4518610","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4518610","url":null,"abstract":"Automatic phonetic reconstruction of medical dictations from non- literal and automatically recognized speech transcripts leads to closer-to-literal transcripts for training. In this paper, we introduce an extended alignment method assessing multiple levels of text segmentation and show how open issues like wrong segmentation in the recognized transcript can be resolved. Furthermore, the effect of context-dependent reconstruction and the phonetic similarity threshold on the quality of the reconstructed transcription is measured. Experiments show an increase in precision between 0.7% and 4.7% absolute without loss in recall for the combined system incorporating all of these techniques in comparison to the system in the previous work.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114279053","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 : 2008-05-12DOI: 10.1109/ICASSP.2008.4517935
G. Kannan, A. Milani, I. Panahi
Periodic signals (since they can be easily predicted) can be canceled much more effectively when compared to non- periodic/stochastic signals. A large class of acoustic noise sources have an underlying periodic process that generates a periodic noise component, and thus the acoustic noise can in general be modeled as the sum of a periodic signal and a random signal (usually a background noise). In this paper we present the idea that separating the acoustic noise into periodic and random noise components and doing separate active noise control(ANC) for each tends to increase the over-all noise attenuation level (NAL). Formulae for the exact improvement in noise attenuation levels are derived. A novel signal separation and noise cancelation scheme based on adaptive filtering is developed and its effectiveness is shown for several periodic signal in white noise cases.
{"title":"Active noise control of noisy periodic signals using signal separation","authors":"G. Kannan, A. Milani, I. Panahi","doi":"10.1109/ICASSP.2008.4517935","DOIUrl":"https://doi.org/10.1109/ICASSP.2008.4517935","url":null,"abstract":"Periodic signals (since they can be easily predicted) can be canceled much more effectively when compared to non- periodic/stochastic signals. A large class of acoustic noise sources have an underlying periodic process that generates a periodic noise component, and thus the acoustic noise can in general be modeled as the sum of a periodic signal and a random signal (usually a background noise). In this paper we present the idea that separating the acoustic noise into periodic and random noise components and doing separate active noise control(ANC) for each tends to increase the over-all noise attenuation level (NAL). Formulae for the exact improvement in noise attenuation levels are derived. A novel signal separation and noise cancelation scheme based on adaptive filtering is developed and its effectiveness is shown for several periodic signal in white noise cases.","PeriodicalId":333742,"journal":{"name":"2008 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114294441","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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