Pub Date : 2014-05-04DOI: 10.1109/ICASSP.2014.6853709
Matthew O'Connor, W. Kleijn
Advances in hardware and communication technology make distributed sound acquisition increasingly attractive. We describe a distributed beamforming method based on the diffusion adaptation paradigm. In contrast to existing distributed beamforming methods, the method does not impose conditions on the topology or the structure of the network nor does it require knowledge of the noise co-variance matrix. The algorithm can continuously track changes in the noise covariance matrix, making it suitable for a practical, dynamic environment. It will typically perform one iteration per signal sample, limiting communication requirements. Our experiments confirm the effectiveness of the method.
{"title":"Diffusion-based distributed MVDR beamformer","authors":"Matthew O'Connor, W. Kleijn","doi":"10.1109/ICASSP.2014.6853709","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853709","url":null,"abstract":"Advances in hardware and communication technology make distributed sound acquisition increasingly attractive. We describe a distributed beamforming method based on the diffusion adaptation paradigm. In contrast to existing distributed beamforming methods, the method does not impose conditions on the topology or the structure of the network nor does it require knowledge of the noise co-variance matrix. The algorithm can continuously track changes in the noise covariance matrix, making it suitable for a practical, dynamic environment. It will typically perform one iteration per signal sample, limiting communication requirements. Our experiments confirm the effectiveness of the method.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"29 1","pages":"810-814"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"84506096","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6854545
A. Balavoine, C. Rozell, J. Romberg
Recovering static signals from compressed measurements is an important problem that has been extensively studied in modern signal processing. However, only recently have methods been proposed to tackle the problem of recovering a time-varying sequence from streaming online compressed measurements. In this paper, we study the capacity of the standard iterative soft-thresholding algorithm (ISTA) to perform this task in real-time. In previous work, ISTA has been shown to recover static sparse signals. The present paper demonstrates its ability to perform this recovery online in the dynamical setting where measurements are constantly streaming. Our analysis shows that the ℓ2-distance between the output and the target signal decays according to a linear rate, and is supported by simulations on synthetic and real data.
{"title":"Iterative soft-thresholding for time-varying signal recovery","authors":"A. Balavoine, C. Rozell, J. Romberg","doi":"10.1109/ICASSP.2014.6854545","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854545","url":null,"abstract":"Recovering static signals from compressed measurements is an important problem that has been extensively studied in modern signal processing. However, only recently have methods been proposed to tackle the problem of recovering a time-varying sequence from streaming online compressed measurements. In this paper, we study the capacity of the standard iterative soft-thresholding algorithm (ISTA) to perform this task in real-time. In previous work, ISTA has been shown to recover static sparse signals. The present paper demonstrates its ability to perform this recovery online in the dynamical setting where measurements are constantly streaming. Our analysis shows that the ℓ2-distance between the output and the target signal decays according to a linear rate, and is supported by simulations on synthetic and real data.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"20 1","pages":"4958-4962"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"84860350","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6855198
R. Bernardini, R. Rinaldo
Physically Unclonable Functions (PUFs) have been recently proposed as a way to include, in chips, functions that can act as “fingerprints” of the chip, to be used in applications like chip authentication (strong PUF) or private ID generation (weak PUF). Most of the schemes proposed for weak PUF exploit helper data to make the PUF more reliable. The use of an helper, however, has some drawbacks such as complexity and the introduction of a possible attack point. In this manuscript we propose the use of a new type of weak PUF called helper-less PUFs (H-PUF) that does not require the use of helper data. We analyze theoretically the problem of designing an H-PUF and use the result to show how an H-PUF can be built. We also show that the proposed PUF, albeit being of weak type, can also be used for chip authentication.
{"title":"Helper-less physically unclonable functions and chip authentication","authors":"R. Bernardini, R. Rinaldo","doi":"10.1109/ICASSP.2014.6855198","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6855198","url":null,"abstract":"Physically Unclonable Functions (PUFs) have been recently proposed as a way to include, in chips, functions that can act as “fingerprints” of the chip, to be used in applications like chip authentication (strong PUF) or private ID generation (weak PUF). Most of the schemes proposed for weak PUF exploit helper data to make the PUF more reliable. The use of an helper, however, has some drawbacks such as complexity and the introduction of a possible attack point. In this manuscript we propose the use of a new type of weak PUF called helper-less PUFs (H-PUF) that does not require the use of helper data. We analyze theoretically the problem of designing an H-PUF and use the result to show how an H-PUF can be built. We also show that the proposed PUF, albeit being of weak type, can also be used for chip authentication.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"186 1","pages":"8193-8197"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85094234","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6854343
G. Chardon
We introduce a generalization of the MUSIC algorithm to treat block-sparse signals in a multi-measurement vector framework. We show, through theoretical analysis and numerical experiments, that the requirements in terms of number of snapshots and number of measurements depend not only on the sparsity and on the size of the blocks, but also on the rank of the matrices of coefficients for each block. We apply this algorithm to the localization of directive sources, which can be modeled by block-sparsity in a dictionary of multipoles, and show that it compares favorably to a greedy approach based on the same model.
{"title":"A block-sparse music algorithm for the localization and the identification of directive sources","authors":"G. Chardon","doi":"10.1109/ICASSP.2014.6854343","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854343","url":null,"abstract":"We introduce a generalization of the MUSIC algorithm to treat block-sparse signals in a multi-measurement vector framework. We show, through theoretical analysis and numerical experiments, that the requirements in terms of number of snapshots and number of measurements depend not only on the sparsity and on the size of the blocks, but also on the rank of the matrices of coefficients for each block. We apply this algorithm to the localization of directive sources, which can be modeled by block-sparsity in a dictionary of multipoles, and show that it compares favorably to a greedy approach based on the same model.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"1 1","pages":"3953-3957"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85237092","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6854368
K. Yao, Baolin Peng, G. Zweig, Dong Yu, Xiaolong Li, Feng Gao
Recurrent neural networks (RNNs) have recently produced record setting performance in language modeling and word-labeling tasks. In the word-labeling task, the RNN is used analogously to the more traditional conditional random field (CRF) to assign a label to each word in an input sequence, and has been shown to significantly outperform CRFs. In contrast to CRFs, RNNs operate in an online fashion to assign labels as soon as a word is seen, rather than after seeing the whole word sequence. In this paper, we show that the performance of an RNN tagger can be significantly improved by incorporating elements of the CRF model; specifically, the explicit modeling of output-label dependencies with transition features, its global sequence-level objective function, and offline decoding. We term the resulting model a “recurrent conditional random field” and demonstrate its effectiveness on the ATIS travel domain dataset and a variety of web-search language understanding datasets.
{"title":"Recurrent conditional random field for language understanding","authors":"K. Yao, Baolin Peng, G. Zweig, Dong Yu, Xiaolong Li, Feng Gao","doi":"10.1109/ICASSP.2014.6854368","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854368","url":null,"abstract":"Recurrent neural networks (RNNs) have recently produced record setting performance in language modeling and word-labeling tasks. In the word-labeling task, the RNN is used analogously to the more traditional conditional random field (CRF) to assign a label to each word in an input sequence, and has been shown to significantly outperform CRFs. In contrast to CRFs, RNNs operate in an online fashion to assign labels as soon as a word is seen, rather than after seeing the whole word sequence. In this paper, we show that the performance of an RNN tagger can be significantly improved by incorporating elements of the CRF model; specifically, the explicit modeling of output-label dependencies with transition features, its global sequence-level objective function, and offline decoding. We term the resulting model a “recurrent conditional random field” and demonstrate its effectiveness on the ATIS travel domain dataset and a variety of web-search language understanding datasets.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"1 1","pages":"4077-4081"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85591281","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6853779
D. Brown, Radu David
A distributed coherent transmission scheme in which a cluster of cooperative transmitters form a beam toward an intended receiver while directing nulls at a number of other “protected” receivers is considered. The receivers coordinate the transmissions by estimating the channels and providing feedback to the transmit cluster to facilitate coherent transmission. Since the effective channels including carrier phase and frequency offsets are time-varying, two tracking schemes are compared: (i) “local tracking” where each receiver independently tracks its own channels from the transmit cluster and (ii) “unified tracking” where one receiver (or transmitter) tracks all of the channels in the system. The results show that, while beamforming performance is effectively unchanged, nullforming performance can be improved with unified tracking, especially over short prediction intervals and for larger networks.
{"title":"Receiver-coordinated distributed transmit nullforming with local and unified tracking","authors":"D. Brown, Radu David","doi":"10.1109/ICASSP.2014.6853779","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853779","url":null,"abstract":"A distributed coherent transmission scheme in which a cluster of cooperative transmitters form a beam toward an intended receiver while directing nulls at a number of other “protected” receivers is considered. The receivers coordinate the transmissions by estimating the channels and providing feedback to the transmit cluster to facilitate coherent transmission. Since the effective channels including carrier phase and frequency offsets are time-varying, two tracking schemes are compared: (i) “local tracking” where each receiver independently tracks its own channels from the transmit cluster and (ii) “unified tracking” where one receiver (or transmitter) tracks all of the channels in the system. The results show that, while beamforming performance is effectively unchanged, nullforming performance can be improved with unified tracking, especially over short prediction intervals and for larger networks.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"29 1","pages":"1160-1164"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85890880","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6853636
N. Ksairi, P. Ciblat, C. Martret
We address multiuser power and bandwidth allocation for Orthogonal Frequency Division Multiple Access (OFDMA) networks employing a Type-II Hybrid Automatic Repeat reQuest (HARQ) mechanism, practical Coding and Modulation Schemes (MCSs) and Bit Interleaved Coded Modulation (BICM). The problem is formulated as minimizing the sum power required to satisfy a goodput constraint for each link while its transmit power does not exceed a certain level. Assuming statistical Channel State Information (CSI), we propose an algorithm to compute the corresponding optimal resource allocation and a practical selection of the MCSs that significantly boosts the proposed algorithm.
{"title":"Optimal resource allocation for type-II HARQ based OFDMA ad hoc networks under individual rate and power constraints","authors":"N. Ksairi, P. Ciblat, C. Martret","doi":"10.1109/ICASSP.2014.6853636","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6853636","url":null,"abstract":"We address multiuser power and bandwidth allocation for Orthogonal Frequency Division Multiple Access (OFDMA) networks employing a Type-II Hybrid Automatic Repeat reQuest (HARQ) mechanism, practical Coding and Modulation Schemes (MCSs) and Bit Interleaved Coded Modulation (BICM). The problem is formulated as minimizing the sum power required to satisfy a goodput constraint for each link while its transmit power does not exceed a certain level. Assuming statistical Channel State Information (CSI), we propose an algorithm to compute the corresponding optimal resource allocation and a practical selection of the MCSs that significantly boosts the proposed algorithm.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"26 1","pages":"449-453"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85901312","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6854180
T. V. K. Chaitanya, D. Danev, E. Larsson
We propose a nonlinear signal space diversity (SSD) precoding technique that produces transmit signals that have constant envelope (CE) in discrete time, resulting in low peak-to-average power ratio (PAPR) waveforms after pulse-shape filtering. We propose two methods for construction of CE signal set. While the proposed CE-SSD scheme is inferior to the conventional SSD designs in terms of coding gain performance, it performs better in terms of overall power efficiency because of the reduced back-off requirement of the power amplifier (PA).
{"title":"Constant envelope signal space diversity","authors":"T. V. K. Chaitanya, D. Danev, E. Larsson","doi":"10.1109/ICASSP.2014.6854180","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854180","url":null,"abstract":"We propose a nonlinear signal space diversity (SSD) precoding technique that produces transmit signals that have constant envelope (CE) in discrete time, resulting in low peak-to-average power ratio (PAPR) waveforms after pulse-shape filtering. We propose two methods for construction of CE signal set. While the proposed CE-SSD scheme is inferior to the conventional SSD designs in terms of coding gain performance, it performs better in terms of overall power efficiency because of the reduced back-off requirement of the power amplifier (PA).","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"9 1","pages":"3147-3151"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85988367","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6854609
J. Lipor, Sajid Ahmed, Mohamed-Slim Alouini
In multiple-input multiple-output (MIMO) radar setting, it is often desirable to design correlated waveforms such that power is transmitted only to a given set of locations, a process known as beampattern design. To design desired beam-pattern, current research uses iterative algorithms, first to synthesize the waveform covariance matrix, R, then to design the actual waveforms to realize R. In contrast to this, we present a closed form method to design R that exploits discrete Fourier transform and Toeplitz matrix. The resulting covariance matrix fulfills the practical constraints and performance is similar to that of iterative methods. Next, we present a radar architecture for the desired beampattern that does not require the synthesis of covariance matrix nor the design of correlated waveforms.
{"title":"Closed form fourier-based transmit beamforming for MIMO radar","authors":"J. Lipor, Sajid Ahmed, Mohamed-Slim Alouini","doi":"10.1109/ICASSP.2014.6854609","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854609","url":null,"abstract":"In multiple-input multiple-output (MIMO) radar setting, it is often desirable to design correlated waveforms such that power is transmitted only to a given set of locations, a process known as beampattern design. To design desired beam-pattern, current research uses iterative algorithms, first to synthesize the waveform covariance matrix, R, then to design the actual waveforms to realize R. In contrast to this, we present a closed form method to design R that exploits discrete Fourier transform and Toeplitz matrix. The resulting covariance matrix fulfills the practical constraints and performance is similar to that of iterative methods. Next, we present a radar architecture for the desired beampattern that does not require the synthesis of covariance matrix nor the design of correlated waveforms.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"42 1","pages":"5272-5276"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"76834477","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 : 2014-05-04DOI: 10.1109/ICASSP.2014.6854897
Ivan Dokmanić, I. Tashev
Depth imaging is commonly based on light. For example, LIDAR and Kinect use infrared light, while stereo cameras use visible light. These systems require hardware operating at high sampling frequencies, precise calibration, and they dissipate significant power. In this paper, we investigate the potential of ultrasound for image and depth acquisition, with applications to human-computer interaction and skeletal tracking in mind. We use a loudspeaker array and a microphone array to sense the scene. We discuss a technique for offline loudspeaker beamforming (commonly used for microphone beamforming) which enables us to significantly increase the frame rate. Further, we propose a sound-source-localization-based method for computing the depth image, giving a substantial improvement over the näıve time-of-flight approach. We designed inexpensive hardware with eight elements per array to obtain both the depth and the intensity images. Even with this limited number of transducers we obtain promising experimental results.
{"title":"Hardware and algorithms for ultrasonic depth imaging","authors":"Ivan Dokmanić, I. Tashev","doi":"10.1109/ICASSP.2014.6854897","DOIUrl":"https://doi.org/10.1109/ICASSP.2014.6854897","url":null,"abstract":"Depth imaging is commonly based on light. For example, LIDAR and Kinect use infrared light, while stereo cameras use visible light. These systems require hardware operating at high sampling frequencies, precise calibration, and they dissipate significant power. In this paper, we investigate the potential of ultrasound for image and depth acquisition, with applications to human-computer interaction and skeletal tracking in mind. We use a loudspeaker array and a microphone array to sense the scene. We discuss a technique for offline loudspeaker beamforming (commonly used for microphone beamforming) which enables us to significantly increase the frame rate. Further, we propose a sound-source-localization-based method for computing the depth image, giving a substantial improvement over the näıve time-of-flight approach. We designed inexpensive hardware with eight elements per array to obtain both the depth and the intensity images. Even with this limited number of transducers we obtain promising experimental results.","PeriodicalId":6545,"journal":{"name":"2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"19 1","pages":"6702-6706"},"PeriodicalIF":0.0,"publicationDate":"2014-05-04","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"76897760","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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