Pub Date : 2010-12-10DOI: 10.1109/MMSP.2010.5661989
Weig-Ge Chen, Zhengyou Zhang
It has been proven that spatial audio enhances the realism of sound for teleconferencing. Previously, solutions have been proposed for multiparty conferencing where each remote participant is assumed to have his/her own microphone, and for conferencing between two rooms where the microphones in one room are connected to the equal number of loudspeakers in the other room. Either approach has its limitations. Hence, we propose a new scheme to improve stereophonic conferencing experience through an innovative use of microphone arrays. Instead of operating in the default mode where a single channel is produced using spatial filtering, we propose to transmit all channels forming a collection of spatial samples of the sound field. Those samples are warped appropriately at the remote site, and are spatialized together with audio streams from other remote sites if any, to produce the perception of a virtual sound field. Real-world audio samples are provided to showcase the proposed technique. The informal listening test shows that majority of the users prefer the new experience.
{"title":"Enhancing stereophonic teleconferencing with microphone arrays through sound field warping","authors":"Weig-Ge Chen, Zhengyou Zhang","doi":"10.1109/MMSP.2010.5661989","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5661989","url":null,"abstract":"It has been proven that spatial audio enhances the realism of sound for teleconferencing. Previously, solutions have been proposed for multiparty conferencing where each remote participant is assumed to have his/her own microphone, and for conferencing between two rooms where the microphones in one room are connected to the equal number of loudspeakers in the other room. Either approach has its limitations. Hence, we propose a new scheme to improve stereophonic conferencing experience through an innovative use of microphone arrays. Instead of operating in the default mode where a single channel is produced using spatial filtering, we propose to transmit all channels forming a collection of spatial samples of the sound field. Those samples are warped appropriately at the remote site, and are spatialized together with audio streams from other remote sites if any, to produce the perception of a virtual sound field. Real-world audio samples are provided to showcase the proposed technique. The informal listening test shows that majority of the users prefer the new experience.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"12 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116206366","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5662004
Amit Bleiweiss, M. Werman
We present a new solution for real-time head pose estimation. The key to our method is a model-based approach based on the fusion of color and time-of-flight depth data. Our method has several advantages over existing head-pose estimation solutions. It requires no initial setup or knowledge of a pre-built model or training data. The use of additional depth data leads to a robust solution, while maintaining real-time performance. The method outperforms the state-of-the art in several experiments using extreme situations such as sudden changes in lighting, large rotations, and fast motion.
{"title":"Robust head pose estimation by fusing time-of-flight depth and color","authors":"Amit Bleiweiss, M. Werman","doi":"10.1109/MMSP.2010.5662004","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5662004","url":null,"abstract":"We present a new solution for real-time head pose estimation. The key to our method is a model-based approach based on the fusion of color and time-of-flight depth data. Our method has several advantages over existing head-pose estimation solutions. It requires no initial setup or knowledge of a pre-built model or training data. The use of additional depth data leads to a robust solution, while maintaining real-time performance. The method outperforms the state-of-the art in several experiments using extreme situations such as sudden changes in lighting, large rotations, and fast motion.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"138 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133983252","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5662046
H. Benhabiles, G. Lavoué, Jean-Philippe Vandeborre, M. Daoudi
In this paper we present a subjective quality assessment experiment for 3D-mesh segmentation. For this end, we carefully designed a protocol with respect to several factors namely the rendering conditions, the possible interactions, the rating range, and the number of human subjects. To carry out the subjective experiment, more than 40 human observers have rated a set of 250 segmentation results issued from various algorithms. The obtained Mean Opinion Scores, which represent the human subjects' point of view toward the quality of each segmentation, have then been used to evaluate both the quality of automatic segmentation algorithms and the quality of similarity metrics used in recent mesh segmentation benchmarking systems.
{"title":"A subjective experiment for 3D-mesh segmentation evaluation","authors":"H. Benhabiles, G. Lavoué, Jean-Philippe Vandeborre, M. Daoudi","doi":"10.1109/MMSP.2010.5662046","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5662046","url":null,"abstract":"In this paper we present a subjective quality assessment experiment for 3D-mesh segmentation. For this end, we carefully designed a protocol with respect to several factors namely the rendering conditions, the possible interactions, the rating range, and the number of human subjects. To carry out the subjective experiment, more than 40 human observers have rated a set of 250 segmentation results issued from various algorithms. The obtained Mean Opinion Scores, which represent the human subjects' point of view toward the quality of each segmentation, have then been used to evaluate both the quality of automatic segmentation algorithms and the quality of similarity metrics used in recent mesh segmentation benchmarking systems.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"684 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132868006","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5662023
Wen Yang, O. Au, Jingjing Dai, Feng Zou, Chao Pang, Yu Liu
Motion estimation as well as the corresponding motion compensation is a core part of modern video coding standards, which highly improves the compression efficiency. On the other hand, motion information takes considerable portion of compressed bit stream, especially in low bit rate situation. In this paper, an efficient motion vector prediction algorithm is proposed to minimize the bits used for coding the motion information. First, a possible motion vector predictor (MVP) candidate set (CS) including several scaled spatial and temporal predictors is defined. To increase the diversity of predictors, the spatial predictor is adaptively changed based on current distribution of neighboring motion vectors. After that, adaptive template matching technique is applied to remove non-effective predictors from the CS so that the bits used for the MVP index can be significantly reduced. As the final MVP is chosen based on minimum motion vector difference criterion, a guessing strategy is further introduced so that in some situations the bits consumed by signaling the MVP index to the decoder can be totally omitted. The experimental results indicate that the proposed method can achieve an average bit rate reduction of 5.9% compared with the H.264 standard.
{"title":"Motion vector coding algorithm based on adaptive template matching","authors":"Wen Yang, O. Au, Jingjing Dai, Feng Zou, Chao Pang, Yu Liu","doi":"10.1109/MMSP.2010.5662023","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5662023","url":null,"abstract":"Motion estimation as well as the corresponding motion compensation is a core part of modern video coding standards, which highly improves the compression efficiency. On the other hand, motion information takes considerable portion of compressed bit stream, especially in low bit rate situation. In this paper, an efficient motion vector prediction algorithm is proposed to minimize the bits used for coding the motion information. First, a possible motion vector predictor (MVP) candidate set (CS) including several scaled spatial and temporal predictors is defined. To increase the diversity of predictors, the spatial predictor is adaptively changed based on current distribution of neighboring motion vectors. After that, adaptive template matching technique is applied to remove non-effective predictors from the CS so that the bits used for the MVP index can be significantly reduced. As the final MVP is chosen based on minimum motion vector difference criterion, a guessing strategy is further introduced so that in some situations the bits consumed by signaling the MVP index to the decoder can be totally omitted. The experimental results indicate that the proposed method can achieve an average bit rate reduction of 5.9% compared with the H.264 standard.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117337709","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5662058
M. Poderico, S. Parrilli, G. Poggi, L. Verdoliva
In this work we propose a modified version of the BM3D algorithm recently introduced by Dabov et al. [1] for the denoising of images corrupted by additive white Gaussian noise. The original technique performs a multipoint filtering, where the nonlocal approach is combined with the wavelet shrinkage of a 3D cube composed by similar patches collected by means of block-matching. Our improvement concerns the thresholding of wavelet coefficients, which are subject to a different shrinkage depending on their level of sparsity. The modified algorithm is more robust with respect to block matching errors, especially when noise is high, as proved by experimental results on a large set of natural images.
{"title":"Sigmoid shrinkage for BM3D denoising algorithm","authors":"M. Poderico, S. Parrilli, G. Poggi, L. Verdoliva","doi":"10.1109/MMSP.2010.5662058","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5662058","url":null,"abstract":"In this work we propose a modified version of the BM3D algorithm recently introduced by Dabov et al. [1] for the denoising of images corrupted by additive white Gaussian noise. The original technique performs a multipoint filtering, where the nonlocal approach is combined with the wavelet shrinkage of a 3D cube composed by similar patches collected by means of block-matching. Our improvement concerns the thresholding of wavelet coefficients, which are subject to a different shrinkage depending on their level of sparsity. The modified algorithm is more robust with respect to block matching errors, especially when noise is high, as proved by experimental results on a large set of natural images.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"50 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115986177","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5662032
Mohsen Akbari, F. Labeau
In this paper, we propose a method for reconstructing the output of an Oversampled Filter Bank (OFB) when instantaneous erasures happen in the sub-band domain. Instantaneous erasure is defined as a situation where the erasure pattern changes in each time instance. This definition differs from the type of erasure usually defined in literature, where e erasures means that e channels of the OFB are off and do not work at all. This new definition is more realistic and increases the flexibility and resilience of the OFB in combating the erasures. Additionally, similar to puncturing, the same idea can be used in an erasure-free channel to reconstruct the output, when sub-band samples are discarded intentionally in order to change the code rate. In this paper we also derive the sufficient conditions that should be met by the OFB in order for the proposed reconstruction method to work. Based on that, eventually we suggest a general form for the OFBs which are robust to this type of erasure.
{"title":"Recovering the output of an OFB in the case of instantaneous erasures in sub-band domain","authors":"Mohsen Akbari, F. Labeau","doi":"10.1109/MMSP.2010.5662032","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5662032","url":null,"abstract":"In this paper, we propose a method for reconstructing the output of an Oversampled Filter Bank (OFB) when instantaneous erasures happen in the sub-band domain. Instantaneous erasure is defined as a situation where the erasure pattern changes in each time instance. This definition differs from the type of erasure usually defined in literature, where e erasures means that e channels of the OFB are off and do not work at all. This new definition is more realistic and increases the flexibility and resilience of the OFB in combating the erasures. Additionally, similar to puncturing, the same idea can be used in an erasure-free channel to reconstruct the output, when sub-band samples are discarded intentionally in order to change the code rate. In this paper we also derive the sufficient conditions that should be met by the OFB in order for the proposed reconstruction method to work. Based on that, eventually we suggest a general form for the OFBs which are robust to this type of erasure.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117099383","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5661999
Mashhour Solh, G. Al-Regib
In this paper we propose a new approach for disocclusion removal in depth image-based rendering (DIBR) for 3D-TV. The new approach, Hierarchical Hole-Filling (HHF), eliminates the need for any preprocessing of the depth map. HHF uses a pyramid like approach to estimate the hole pixels from lower resolution estimates of the 3D wrapped image. The lower resolution estimates involves a pseudo zero canceling plus Gaussian filtering of the wrapped image. Then starting backwards from the lowest resolution hole-free estimate in the pyramid, we interpolate and use the pixel values to fill in the hole in the higher up resolution image. The procedure is repeated until the estimated image is hole-free. Experimental results show that HHF yields virtual images that are free of any geometric distortions, which is not the case in other algorithms that preprocess the depth map. Experiments has also shown that unlike previous DIBR techniques, HHF is not sensitive to depth maps with high percentage of bad matching pixels.
{"title":"Hierarchical Hole-Filling(HHF): Depth image based rendering without depth map filtering for 3D-TV","authors":"Mashhour Solh, G. Al-Regib","doi":"10.1109/MMSP.2010.5661999","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5661999","url":null,"abstract":"In this paper we propose a new approach for disocclusion removal in depth image-based rendering (DIBR) for 3D-TV. The new approach, Hierarchical Hole-Filling (HHF), eliminates the need for any preprocessing of the depth map. HHF uses a pyramid like approach to estimate the hole pixels from lower resolution estimates of the 3D wrapped image. The lower resolution estimates involves a pseudo zero canceling plus Gaussian filtering of the wrapped image. Then starting backwards from the lowest resolution hole-free estimate in the pyramid, we interpolate and use the pixel values to fill in the hole in the higher up resolution image. The procedure is repeated until the estimated image is hole-free. Experimental results show that HHF yields virtual images that are free of any geometric distortions, which is not the case in other algorithms that preprocess the depth map. Experiments has also shown that unlike previous DIBR techniques, HHF is not sensitive to depth maps with high percentage of bad matching pixels.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"45 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123242636","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5661988
Martin Rothbucher, Tim Habigt, Johannes Feldmaier, K. Diepold
This paper describes an integration of a Head Related Transfer Function (HRTF)-based 3D sound convolution engine into the open-source VoIP conferencing software Mumble. Our system allows to virtually place audio contributions of conference participants to different positions around a listener, which helps to overcome the problem of identifying active speakers in an audio conference. Furthermore, using HRTFs to generate 3D sound in virtual 3D space, the listener is able to make use of the cocktail party effect in order to differentiate between several simultaneously active speakers. As a result intelligibility of communication is increased.
{"title":"Integrating a HRTF-based sound synthesis system into Mumble","authors":"Martin Rothbucher, Tim Habigt, Johannes Feldmaier, K. Diepold","doi":"10.1109/MMSP.2010.5661988","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5661988","url":null,"abstract":"This paper describes an integration of a Head Related Transfer Function (HRTF)-based 3D sound convolution engine into the open-source VoIP conferencing software Mumble. Our system allows to virtually place audio contributions of conference participants to different positions around a listener, which helps to overcome the problem of identifying active speakers in an audio conference. Furthermore, using HRTFs to generate 3D sound in virtual 3D space, the listener is able to make use of the cocktail party effect in order to differentiate between several simultaneously active speakers. As a result intelligibility of communication is increased.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130238545","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5662066
V. Estellers, J. Thiran
In this paper we propose two alternatives to overcome the natural asynchrony of modalities in Audio-Visual Speech Recognition. We first investigate the use of asynchronous statistical models based on Dynamic Bayesian Networks with different levels of asynchrony. We show that audio-visual models should consider asynchrony within word boundaries and not at phoneme level. The second approach to the problem includes an additional processing of the features before being used for recognition. The proposed technique aligns the temporal evolution of the audio and video streams in terms of a speech-recognition system and enables the use of simpler statistical models for classification. On both cases we report experiments with the CUAVE database, showing the improvements obtained with the proposed asynchronous model and feature processing technique compared to traditional systems.
{"title":"Overcoming asynchrony in Audio-Visual Speech Recognition","authors":"V. Estellers, J. Thiran","doi":"10.1109/MMSP.2010.5662066","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5662066","url":null,"abstract":"In this paper we propose two alternatives to overcome the natural asynchrony of modalities in Audio-Visual Speech Recognition. We first investigate the use of asynchronous statistical models based on Dynamic Bayesian Networks with different levels of asynchrony. We show that audio-visual models should consider asynchrony within word boundaries and not at phoneme level. The second approach to the problem includes an additional processing of the features before being used for recognition. The proposed technique aligns the temporal evolution of the audio and video streams in terms of a speech-recognition system and enables the use of simpler statistical models for classification. On both cases we report experiments with the CUAVE database, showing the improvements obtained with the proposed asynchronous model and feature processing technique compared to traditional systems.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130315000","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 : 2010-12-10DOI: 10.1109/MMSP.2010.5662001
A. A. Moghadam, H. Radha
We consider the problem of recovering a signal/image (x) with a k-sparse representation, from hybrid (complex and real), noiseless linear samples (y) using a mixture of complex-valued sparse and real-valued dense projections within a single matrix. The proposed Hybrid Compressed Sensing (HCS) employs the complex-sparse part of the projection matrix to divide the n-dimensional signal (x) into subsets. In turn, each subset of the signal (coefficients) is mapped onto a complex sample of the measurement vector (y). Under a worst-case scenario of such sparsity-induced mapping, when the number of complex sparse measurements is sufficiently large then this mapping leads to the isolation of a significant fraction of the k non-zero coefficients into different complex measurement samples from y. Using a simple property of complex numbers (namely complex phases) one can identify the isolated non-zeros of x. After reducing the effect of the identified non-zero coefficients from the compressive samples, we utilize the real-valued dense submatrix to form a full rank system of equations to recover the signal values in the remaining indices (that are not recovered by the sparse complex projection part). We show that the proposed hybrid approach can recover a k-sparse signal (with high probability) while requiring only m ≈ 3√n/2k real measurements (where each complex sample is counted as two real measurements). We also derive expressions for the optimal mix of complex-sparse and real-dense rows within an HCS projection matrix. Further, in a practical range of sparsity ratio (k/n) suitable for images, the hybrid approach outperforms even the most complex compressed sensing frameworks (namely basis pursuit with dense Gaussian matrices). The theoretical complexity of HCS is less than the complexity of solving a full-rank system of m linear equations. In practice, the complexity can be lower than this bound.
{"title":"Hybrid Compressed Sensing of images","authors":"A. A. Moghadam, H. Radha","doi":"10.1109/MMSP.2010.5662001","DOIUrl":"https://doi.org/10.1109/MMSP.2010.5662001","url":null,"abstract":"We consider the problem of recovering a signal/image (x) with a k-sparse representation, from hybrid (complex and real), noiseless linear samples (y) using a mixture of complex-valued sparse and real-valued dense projections within a single matrix. The proposed Hybrid Compressed Sensing (HCS) employs the complex-sparse part of the projection matrix to divide the n-dimensional signal (x) into subsets. In turn, each subset of the signal (coefficients) is mapped onto a complex sample of the measurement vector (y). Under a worst-case scenario of such sparsity-induced mapping, when the number of complex sparse measurements is sufficiently large then this mapping leads to the isolation of a significant fraction of the k non-zero coefficients into different complex measurement samples from y. Using a simple property of complex numbers (namely complex phases) one can identify the isolated non-zeros of x. After reducing the effect of the identified non-zero coefficients from the compressive samples, we utilize the real-valued dense submatrix to form a full rank system of equations to recover the signal values in the remaining indices (that are not recovered by the sparse complex projection part). We show that the proposed hybrid approach can recover a k-sparse signal (with high probability) while requiring only m ≈ 3√n/2k real measurements (where each complex sample is counted as two real measurements). We also derive expressions for the optimal mix of complex-sparse and real-dense rows within an HCS projection matrix. Further, in a practical range of sparsity ratio (k/n) suitable for images, the hybrid approach outperforms even the most complex compressed sensing frameworks (namely basis pursuit with dense Gaussian matrices). The theoretical complexity of HCS is less than the complexity of solving a full-rank system of m linear equations. In practice, the complexity can be lower than this bound.","PeriodicalId":105774,"journal":{"name":"2010 IEEE International Workshop on Multimedia Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-12-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131158365","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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