Pub Date : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458170
E. Hellerud, U. Svensson
A system for streaming lossless audio with low delay over IP networks is presented. To achieve error robustness, the signal is divided into a base and an enhancement layer when the network is approaching congestion. The base layer is perceptually encoded using a time-varying pre- and postfilter, and this layer is transported using a high priority traffic class in a Differentiated Services (DiffServ) network. The enhancement layer is the difference between the original signal and base layer, and is transmitted using a regular Best Effort traffic class. In our experiments the system delay is just 256 samples, and it can be seen that the layering only introduces moderate amounts of redundancy, while improving the error resilience significantly.
{"title":"Robust Transmission of Lossless Audio with Low Delay over IP Networks","authors":"E. Hellerud, U. Svensson","doi":"10.1109/ISSPIT.2007.4458170","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458170","url":null,"abstract":"A system for streaming lossless audio with low delay over IP networks is presented. To achieve error robustness, the signal is divided into a base and an enhancement layer when the network is approaching congestion. The base layer is perceptually encoded using a time-varying pre- and postfilter, and this layer is transported using a high priority traffic class in a Differentiated Services (DiffServ) network. The enhancement layer is the difference between the original signal and base layer, and is transmitted using a regular Best Effort traffic class. In our experiments the system delay is just 256 samples, and it can be seen that the layering only introduces moderate amounts of redundancy, while improving the error resilience significantly.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"51 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133171627","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458159
F. Awan, S. Marshall, J. Soraghan, M. N. Arbab
This paper compares different robust techniques used for imperceptible watermarking, which are resistant to geometric and signal processing attacks. The dual tree complex wavelet transform (DT CWT) provides a means of producing solutions for robust watermarking. Complex wavelet transforms use Gabor real time filters having the properties of shift invariance and directional selectivity. This produces a considerable reduction in the complexity making the DT CWT an ideal solution for real time watermarking. The discrete cosine transform (DCT) has also been proposed for robust watermarking because of its resistance to geometrical attacks. In this paper a hybrid system comprising the DCT and DT CWT is used in order to produce a more robust technique for watermarking. In the process of embedding a watermark, the properties of a human visual system (HVS) are also considered. This paper compares the watermarking process for DT CWT, DT CWT with DCT and discrete wavelets transform (DWT) in producing robust watermarks. This paper demonstrates the superior performance in the presence of geometric and signal processing attacks of the hybrid DT CWT-DCT technique.
{"title":"Performance of a Hybrid DCT - DT CWT Digital Watermarking against Geometric and Signal Processing Attacks","authors":"F. Awan, S. Marshall, J. Soraghan, M. N. Arbab","doi":"10.1109/ISSPIT.2007.4458159","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458159","url":null,"abstract":"This paper compares different robust techniques used for imperceptible watermarking, which are resistant to geometric and signal processing attacks. The dual tree complex wavelet transform (DT CWT) provides a means of producing solutions for robust watermarking. Complex wavelet transforms use Gabor real time filters having the properties of shift invariance and directional selectivity. This produces a considerable reduction in the complexity making the DT CWT an ideal solution for real time watermarking. The discrete cosine transform (DCT) has also been proposed for robust watermarking because of its resistance to geometrical attacks. In this paper a hybrid system comprising the DCT and DT CWT is used in order to produce a more robust technique for watermarking. In the process of embedding a watermark, the properties of a human visual system (HVS) are also considered. This paper compares the watermarking process for DT CWT, DT CWT with DCT and discrete wavelets transform (DWT) in producing robust watermarks. This paper demonstrates the superior performance in the presence of geometric and signal processing attacks of the hybrid DT CWT-DCT technique.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133214922","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458200
Sabbir U. Ahmad, Andreas Antoniou
A genetic algorithm (GA) for the design of asymmetric FIR filters that would satisfy multiple requirements imposed on the amplitude response and group-delay characteristic is proposed. The GA implements a multiobjective optimization approach for obtaining so-called Pareto-optimal solutions of the problem at hand. Flexibility is introduced in the design by imposing phase linearity only in the passband instead of the entire baseband as in conventional designs. The proposed GA is a specially tailored elitist nondominated sorting genetic algorithm (ENSGA) and it involves a decimal encoding scheme and a multiobjective error formulation based on the amplitude response and passband group delay. Experimental results show that the ENSGA leads to improved amplitude response as well as delay characteristic relative to those achieved by using a state-of-the-art weighted least-squares approach.
{"title":"A Multiobjective Genetic Algorithm for Asymmetric FIR Filters","authors":"Sabbir U. Ahmad, Andreas Antoniou","doi":"10.1109/ISSPIT.2007.4458200","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458200","url":null,"abstract":"A genetic algorithm (GA) for the design of asymmetric FIR filters that would satisfy multiple requirements imposed on the amplitude response and group-delay characteristic is proposed. The GA implements a multiobjective optimization approach for obtaining so-called Pareto-optimal solutions of the problem at hand. Flexibility is introduced in the design by imposing phase linearity only in the passband instead of the entire baseband as in conventional designs. The proposed GA is a specially tailored elitist nondominated sorting genetic algorithm (ENSGA) and it involves a decimal encoding scheme and a multiobjective error formulation based on the amplitude response and passband group delay. Experimental results show that the ENSGA leads to improved amplitude response as well as delay characteristic relative to those achieved by using a state-of-the-art weighted least-squares approach.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"40 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117239301","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458113
Peeranat Thoonsangngam, S. Thainimit, V. Areekul
This paper proposes a new scheme to generate iris codes based on relative measure of local iris texture. The local characteristic of iris texture is analyzed using 2D Gabor wavelets. Twelve Gabor kernels, four frequencies and three orientations, are constructed and convoluted with an iris image. To inherit relationship of local iris texture among pixels, Gabor magnitude and phase of a reference pixel is compared with Gabor magnitudes and phases of the other four pixels. These pixels are located away from the reference pixel by 8timesd pixels, where d=1, 2, ..., 4. Each comparison, a 2-bit primitive iris code is generated. Least significant bit of the primitive code describes how Gabor magnitudes of the two pixels are related. This bit is set to '1' if Gabor magnitude of a reference pixel is less than magnitude of the other pixel, otherwise it is set to '0'. Another bit of the 2-bit primitive code describes relative measure of the obtained phase values. This bit is set to '1' if difference of the obtained phases is within plusmnpi/2 , otherwise it is set to '0'. In our scheme, each pixel is described using an 8-bit iris code. Matching between two iris codes is implemented using a look-up table technique. The table contains a number of matches of the primitive code of the two iris codes. By utilizing the look-up table technique, computational time of our 1:1 matching scheme is 2.2 milliseconds. Equal- Error-Rate (EER) of the proposed system using CASIA1.0 iris database is 0.0003%EER
{"title":"Relative Iris Codes","authors":"Peeranat Thoonsangngam, S. Thainimit, V. Areekul","doi":"10.1109/ISSPIT.2007.4458113","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458113","url":null,"abstract":"This paper proposes a new scheme to generate iris codes based on relative measure of local iris texture. The local characteristic of iris texture is analyzed using 2D Gabor wavelets. Twelve Gabor kernels, four frequencies and three orientations, are constructed and convoluted with an iris image. To inherit relationship of local iris texture among pixels, Gabor magnitude and phase of a reference pixel is compared with Gabor magnitudes and phases of the other four pixels. These pixels are located away from the reference pixel by 8timesd pixels, where d=1, 2, ..., 4. Each comparison, a 2-bit primitive iris code is generated. Least significant bit of the primitive code describes how Gabor magnitudes of the two pixels are related. This bit is set to '1' if Gabor magnitude of a reference pixel is less than magnitude of the other pixel, otherwise it is set to '0'. Another bit of the 2-bit primitive code describes relative measure of the obtained phase values. This bit is set to '1' if difference of the obtained phases is within plusmnpi/2 , otherwise it is set to '0'. In our scheme, each pixel is described using an 8-bit iris code. Matching between two iris codes is implemented using a look-up table technique. The table contains a number of matches of the primitive code of the two iris codes. By utilizing the look-up table technique, computational time of our 1:1 matching scheme is 2.2 milliseconds. Equal- Error-Rate (EER) of the proposed system using CASIA1.0 iris database is 0.0003%EER","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114258359","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458055
Angel R Kanchev, A. Popova
A system for development of authoring, editing, analyzing and compressing / decompressing algorithms for video and audio is presented. The system uses DirectShow, QuickTime and its own MPEG-4 framework. It supports multi-language and multi-platform (native, Java and .Net) plug-in system. The system's core as well as its GUI (graphical user interface) gives access to any data in the process of authoring, editing and analyzing. The system gives extensive access to compressed and non-compressed data in many of its representations. This way the system is suitable for developers of multimedia tools or for specialists who need to modify data that is inaccessible in other editing tools.
{"title":"Development System for Video and Audio Algorithms","authors":"Angel R Kanchev, A. Popova","doi":"10.1109/ISSPIT.2007.4458055","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458055","url":null,"abstract":"A system for development of authoring, editing, analyzing and compressing / decompressing algorithms for video and audio is presented. The system uses DirectShow, QuickTime and its own MPEG-4 framework. It supports multi-language and multi-platform (native, Java and .Net) plug-in system. The system's core as well as its GUI (graphical user interface) gives access to any data in the process of authoring, editing and analyzing. The system gives extensive access to compressed and non-compressed data in many of its representations. This way the system is suitable for developers of multimedia tools or for specialists who need to modify data that is inaccessible in other editing tools.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125277317","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458148
A. Fam, I. Sarkar, T. Poonnen
A computationally efficient mismatched filter comprised of a matched filter in cascade with a multi-stage filter based on v is proposed. For this approach to work, the autocorrelation of the given code has to satisfy certain conditions that are derived in this work. If in addition, the sidelobes are sparse and of small integer values, such as in Barker codes and Huffman sequences, then the proposed filters are shown to be computationally very efficient. When implemented in VLSI, they require significantly smaller chip area and less power compared to the length-optimal filters achieving comparable sidelobe suppression.
{"title":"Computationally Efficient Mismatched Filters Based on Sidelobe Inversion","authors":"A. Fam, I. Sarkar, T. Poonnen","doi":"10.1109/ISSPIT.2007.4458148","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458148","url":null,"abstract":"A computationally efficient mismatched filter comprised of a matched filter in cascade with a multi-stage filter based on v is proposed. For this approach to work, the autocorrelation of the given code has to satisfy certain conditions that are derived in this work. If in addition, the sidelobes are sparse and of small integer values, such as in Barker codes and Huffman sequences, then the proposed filters are shown to be computationally very efficient. When implemented in VLSI, they require significantly smaller chip area and less power compared to the length-optimal filters achieving comparable sidelobe suppression.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125863224","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458143
Bin Zhan, Baochun Hou, Reza Sotudeh
The rate distortion optimization (RDO) enabled mode decision (MD) is one of the most important techniques introduced by H.264/AVC. By adopting the exhaustive calculation of rate distortion, the optimal MD enhances the video encoding quality. However, the computational complexity is significantly increased, which is a key challenge for real-time and low power consumption applications. This paper presents a new fast MD algorithm for highly efficient H.264/AVC encoder. The proposed algorithm employs a dynamic group of candidate inter/intra modes to reduce the computational cost. In order to minimize the performance loss incurred by improper mode selection for the previously encoded frames, an adaptive adjustment scheme based on the undulation of bitrate and PSNR is suggested. Experimental results show that the proposed algorithm reduces the encoding time by 35% on average, and the loss of PSNR is usually limited in 0.1 dB with less than 1% increase of bitrate.
{"title":"An Efficient Mode Decision Algorithm Based on Dynamic Grouping and Adaptive Adjustment for H.264/AVC","authors":"Bin Zhan, Baochun Hou, Reza Sotudeh","doi":"10.1109/ISSPIT.2007.4458143","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458143","url":null,"abstract":"The rate distortion optimization (RDO) enabled mode decision (MD) is one of the most important techniques introduced by H.264/AVC. By adopting the exhaustive calculation of rate distortion, the optimal MD enhances the video encoding quality. However, the computational complexity is significantly increased, which is a key challenge for real-time and low power consumption applications. This paper presents a new fast MD algorithm for highly efficient H.264/AVC encoder. The proposed algorithm employs a dynamic group of candidate inter/intra modes to reduce the computational cost. In order to minimize the performance loss incurred by improper mode selection for the previously encoded frames, an adaptive adjustment scheme based on the undulation of bitrate and PSNR is suggested. Experimental results show that the proposed algorithm reduces the encoding time by 35% on average, and the loss of PSNR is usually limited in 0.1 dB with less than 1% increase of bitrate.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125182195","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458066
F. Z. Merli, Matteo Vitetta, Xiaodong Wang
We derive a novel Bayesian algorithm for multiuser detection in the uplink of a multiple-input multiple-output (MIMO) orthogonal frequency division multiplexing (OFDM) system employing stacked space-time block codes, such as the stacked Alamouti code with two transmit antennas, and a stacked quasi-orthogonal code with four transmit antennas. The proposed technique accomplishes joint estimation of the carrier frequency offset, phase noise, channel impulse response and data of each active user. Its derivation relies on the specific structure of the transmitted signal and on efficient Markov chain Monte Carlo (MCMC) methods. Simulation results evidence the robustness of the proposed algorithm.
{"title":"A Bayesian Multiuser Detector for MIMO-OFDM Systems Affected by Multipath Fading, Carrier Frequency Offset and Phase Noise","authors":"F. Z. Merli, Matteo Vitetta, Xiaodong Wang","doi":"10.1109/ISSPIT.2007.4458066","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458066","url":null,"abstract":"We derive a novel Bayesian algorithm for multiuser detection in the uplink of a multiple-input multiple-output (MIMO) orthogonal frequency division multiplexing (OFDM) system employing stacked space-time block codes, such as the stacked Alamouti code with two transmit antennas, and a stacked quasi-orthogonal code with four transmit antennas. The proposed technique accomplishes joint estimation of the carrier frequency offset, phase noise, channel impulse response and data of each active user. Its derivation relies on the specific structure of the transmitted signal and on efficient Markov chain Monte Carlo (MCMC) methods. Simulation results evidence the robustness of the proposed algorithm.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125292511","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458119
F. Keyrouz
It is generally known that sound waves are transformed by the pinnae into sound-pressure signals at the two ear drums. The monaural and inter-aural cues resulting from this process, i.e. spectral cues and interaural phase and intensity differences, are employed by the auditory system in the formation of auditory events. In this context, not only the two pinnae but also the whole head have an important functional role, which is best described as a spatial filtering process. This linear filtering is usually quantified in terms of so-called head-related transfer functions (HRTFs). Motivated by the role of the pinnae to direct and amplify sound, we present a cognitive method for localizing sound sources in a three dimensional space to be deployed in humanoid robotic systems. Using a self-adjusting microphone configuration, the inter-microphone distances dynamically reconfigure in order to optimize the localization accuracy based on the audio signals content. Our new localization system demonstrated high precision 3D sound tracking using only four microphones and enabled a low complexity implementation on the humanoid DSP platform.
{"title":"Automatic Self-Reconfigurating Microphones for Humanoid Dynamic Hearing Environments","authors":"F. Keyrouz","doi":"10.1109/ISSPIT.2007.4458119","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458119","url":null,"abstract":"It is generally known that sound waves are transformed by the pinnae into sound-pressure signals at the two ear drums. The monaural and inter-aural cues resulting from this process, i.e. spectral cues and interaural phase and intensity differences, are employed by the auditory system in the formation of auditory events. In this context, not only the two pinnae but also the whole head have an important functional role, which is best described as a spatial filtering process. This linear filtering is usually quantified in terms of so-called head-related transfer functions (HRTFs). Motivated by the role of the pinnae to direct and amplify sound, we present a cognitive method for localizing sound sources in a three dimensional space to be deployed in humanoid robotic systems. Using a self-adjusting microphone configuration, the inter-microphone distances dynamically reconfigure in order to optimize the localization accuracy based on the audio signals content. Our new localization system demonstrated high precision 3D sound tracking using only four microphones and enabled a low complexity implementation on the humanoid DSP platform.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129774420","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 : 2007-12-01DOI: 10.1109/ISSPIT.2007.4458116
Gamal Fahmy
Super-Resolution image construction has gained increased importance recently. This is due to the demand for resolution enhancement for many imaging applications, as it is much efficient to capture images in a low resolution environment. The Bspline mathematical functions have long been utilized for signal representation. However they have been just recently been used for signal interpolation and zooming. This is due to the fact that they are flexible and provide the best cost/quality trade off relationship. In this paper we present a super-resolution image construction algorithm, where the high frequencies and edges of the high resolution constructed image are solely based on the Bspline signal representation. Mathematical explanation and derivation for the proposed Bspline prediction is analyzed. Several texture images from the Vistex database has been used to test the proposed technique. Extensive simulation results, that have been carried out with the proposed approach on different classes of images and demonstrated its usefulness, are proposed.
{"title":"Bspline based Super-Resolution Construction of Textured Images","authors":"Gamal Fahmy","doi":"10.1109/ISSPIT.2007.4458116","DOIUrl":"https://doi.org/10.1109/ISSPIT.2007.4458116","url":null,"abstract":"Super-Resolution image construction has gained increased importance recently. This is due to the demand for resolution enhancement for many imaging applications, as it is much efficient to capture images in a low resolution environment. The Bspline mathematical functions have long been utilized for signal representation. However they have been just recently been used for signal interpolation and zooming. This is due to the fact that they are flexible and provide the best cost/quality trade off relationship. In this paper we present a super-resolution image construction algorithm, where the high frequencies and edges of the high resolution constructed image are solely based on the Bspline signal representation. Mathematical explanation and derivation for the proposed Bspline prediction is analyzed. Several texture images from the Vistex database has been used to test the proposed technique. Extensive simulation results, that have been carried out with the proposed approach on different classes of images and demonstrated its usefulness, are proposed.","PeriodicalId":299267,"journal":{"name":"2007 IEEE International Symposium on Signal Processing and Information Technology","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2007-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128877919","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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