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Cramér-Rao bound for time reversal active array direction of arrival estimators in multipath environments 多径环境下到达估计时间反转有源阵列方向的cram<s:1> - rao约束
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5496261
F. Foroozan, A. Asif
{"title":"Cramér-Rao bound for time reversal active array direction of arrival estimators in multipath environments","authors":"F. Foroozan, A. Asif","doi":"10.1109/ICASSP.2010.5496261","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496261","url":null,"abstract":"In this paper, we study the Cramér-Rao bound (CRB) for time reversal (TR) based direction of arrival (DOA) estimators operating in a rich multipath environment. Our setup is based on an array of active antennas capable of estimating the range and DOA of a passive target. We derive an analytical expression for the CRB of the TR/DOA estimator and compare it with that of the conventional DOA estimator by expressing the two CRBs in terms of the multipath parameters (multipath's attenuations and delays). Our analytical results are verified by running Ground Penetrating Radar (GPR) simulations using the electromagnetic Finite Difference Time Domain (FDTD) models. Our simulations illustrate the potential of superior performance with gains of up to 15 dB possible with the TR/DOA estimator over the conventional approach.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130334552","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}
引用次数: 10
Countering byzantine attacks in cognitive radio networks 对抗认知无线电网络中的拜占庭式攻击
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5496102
A. Rawat, Priyank Anand, Hao Chen, P. Varshney
{"title":"Countering byzantine attacks in cognitive radio networks","authors":"A. Rawat, Priyank Anand, Hao Chen, P. Varshney","doi":"10.1109/ICASSP.2010.5496102","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496102","url":null,"abstract":"Collaborative (or distributed) spectrum sensing has been shown to have various advantages in terms of spectrum utilization and robustness in cognitive radio networks (CRNs). The data fusion scheme is a key component of collaborative spectrum sensing. We have recently analyzed the problem of Byzantine attacks in CRNs, where malicious users send false sensing data to the fusion center (FC) leading to an increased probability of spectrum sensing error. In this paper, we propose a novel and easy to implement technique to counter Byzantine attacks in CRNs. In this approach, the FC identifies the attackers and removes them from the data fusion process. Our analysis indicates that the proposed scheme is robust against Byzantine attacks and can successfully remove the Byzantines in a short time span.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"93 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126820540","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}
引用次数: 76
The LUNA Spoken Dialogue System: Beyond utterance classification LUNA口语对话系统:超越话语分类
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5494952
Marco Dinarelli, Evgeny A. Stepanov, S. Varges, G. Riccardi
{"title":"The LUNA Spoken Dialogue System: Beyond utterance classification","authors":"Marco Dinarelli, Evgeny A. Stepanov, S. Varges, G. Riccardi","doi":"10.1109/ICASSP.2010.5494952","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5494952","url":null,"abstract":"We present a call routing application for complex problem solving tasks. Up to date work on call routing has been mainly dealing with call-type classification. In this paper we take call routing further: Initial call classification is done in parallel with a robust statistical Spoken Language Understanding module. This is followed by a dialogue to elicit further task-relevant details from the user before passing on the call. The dialogue capability also allows us to obtain clarifications of the initial classifier guess. Based on an evaluation, we show that conducting a dialogue significantly improves upon call routing based on call classification alone. We present both subjective and objective evaluation results of the system according to standard metrics on real users.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"29 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126881552","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}
引用次数: 12
Performance analysis and optimization for ARQ decode-and-forward relaying protocol in fast fading channels 快速衰落信道下ARQ译码转发中继协议性能分析与优化
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5496036
Sangkook Lee, Weifeng Su, S. Batalama, J. Matyjas
{"title":"Performance analysis and optimization for ARQ decode-and-forward relaying protocol in fast fading channels","authors":"Sangkook Lee, Weifeng Su, S. Batalama, J. Matyjas","doi":"10.1109/ICASSP.2010.5496036","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496036","url":null,"abstract":"In this paper, a new analytical approach is developed for the evaluation of the outage probability of decode-and-forward (DF) automatic-repeat-request (ARQ) relaying under packet-rate fading (fast fading) channels. Based on this approach, a closed-form asymptotically tight (as SNR → ∞) approximation of the outage probability is derived, and the diversity order of the DF cooperative ARQ relay scheme is shown to be equal to 2L - 1, where L is the maximum number of ARQ retransmissions. The closed-form expression clearly shows that the achieved diversity is partially due to the DF cooperative relaying and partially due to the fast fading nature of the channels (temporal diversity). Numerical and simulation studies illustrate the theoretical developments.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"32 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129297639","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}
引用次数: 1
Classification of MPEG-2 Transport Stream packet loss visibility MPEG-2传输流丢包可见性分类
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5495279
Jihyun Shin, P. Cosman
{"title":"Classification of MPEG-2 Transport Stream packet loss visibility","authors":"Jihyun Shin, P. Cosman","doi":"10.1109/ICASSP.2010.5495279","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495279","url":null,"abstract":"We classify the visibility of TS (Transport Stream) packet losses for SDTV and HDTV MPEG-2 compressed video streams. TS packet losses can cause various temporal and spatial losses. The visual effect of a TS packet loss depends on many factors, in particular whether the loss causes a whole frame loss or partial frame loss. We develop models for predicting loss visibility for both SDTV and HDTV resolutions for frame loss and partial frame loss cases. We compare the dominant predictive factors and the results for the two resolutions. We achieve more than 85% classification accuracy.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"44 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123852408","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}
引用次数: 4
HMM-based separation of acoustic transfer function for single-channel sound source localization 基于hmm的单通道声源定位声传递函数分离
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5496188
R. Takashima, T. Takiguchi, Y. Ariki
{"title":"HMM-based separation of acoustic transfer function for single-channel sound source localization","authors":"R. Takashima, T. Takiguchi, Y. Ariki","doi":"10.1109/ICASSP.2010.5496188","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496188","url":null,"abstract":"This paper presents a sound source (talker) localization method using only a single microphone, where a HMM (Hidden Markov Model) of clean speech is introduced to estimate the acoustic transfer function from a user's position. The new method is able to carry out this estimation without measuring impulse responses. The frame sequence of the acoustic transfer function is estimated by maximizing the likelihood of training data uttered from a given position, where the cepstral parameters are used to effectively represent useful clean speech. Using the estimated frame sequence data, the GMM (Gaussian Mixture Model) of the acoustic transfer function is created to deal with the influence of a room impulse response. Then, for each test data set, we find a maximum-likelihood GMM from among the estimated GMMs corresponding to each position. The effectiveness of this method has been confirmed by talker localization experiments performed in a room environment.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"234 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124240733","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}
引用次数: 16
Adaptive feedback cancellation in hearing aids using a sinusoidal near-end signal model 使用正弦近端信号模型的助听器自适应反馈抵消
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5496063
Kim Ngo, T. Waterschoot, M. G. Christensen, M. Moonen, S. H. Jensen, J. Wouters
{"title":"Adaptive feedback cancellation in hearing aids using a sinusoidal near-end signal model","authors":"Kim Ngo, T. Waterschoot, M. G. Christensen, M. Moonen, S. H. Jensen, J. Wouters","doi":"10.1109/ICASSP.2010.5496063","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5496063","url":null,"abstract":"Acoustic feedback is a well-known problem in hearing aids, which is caused by the undesired acoustic coupling between the loudspeaker and the microphone. Acoustic feedback limits the maximum amplification that can be used in the hearing aid without making it unstable. The goal of adaptive feedback cancellation (AFC) is to adaptively model the feedback path and estimate the feedback signal, which is then subtracted from the microphone signal. The main problem in identifying the feedback path model is the correlation between the near-end signal and the loudspeaker signal, which is caused by the closed signal loop. A possible solution to this problem is to use the prediction error method (PEM)-based AFC with a linear prediction (LP) model for the near-end signal. In this paper, a modification to the PEM-based AFC is presented where the LP model is replaced by a sinusoidal near-end signal model. More specifically, it is shown that using frequency estimation techniques to estimate the sinusoidal near-end signal model improves the performance of the PEM-based AFC compared to using a LP model. Simulation results for a hearing aid scenario indicate a significant improvement in terms of misadjustment and maximum stable gain increase.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"87 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123635581","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}
引用次数: 16
Multiple Selection Approximation for improved spatio-temporal prediction in video coding 改进视频编码时空预测的多重选择近似
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5495253
Jürgen Seiler, André Kaup
{"title":"Multiple Selection Approximation for improved spatio-temporal prediction in video coding","authors":"Jürgen Seiler, André Kaup","doi":"10.1109/ICASSP.2010.5495253","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495253","url":null,"abstract":"In this contribution, a novel spatio-temporal prediction algorithm for video coding is introduced. This algorithm exploits temporal as well as spatial redundancies for effectively predicting the signal to be encoded. To achieve this, the algorithm operates in two stages. Initially, motion compensated prediction is applied on the block being encoded. Afterwards this preliminary temporal prediction is refined by forming a joint model of the initial predictor and the spatially adjacent already transmitted blocks. The novel algorithm is able to outperform earlier refinement algorithms in speed and prediction quality. Compared to pure motion compensated prediction, the mean data rate can be reduced by up to 15% and up to 1.16 dB gain in PSNR can be achieved for the considered sequences.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"79 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121188026","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}
引用次数: 3
Performance analysis of blind adaptive MIMO receivers 盲自适应MIMO接收机性能分析
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5495966
B. Balasingam, M. Bolic, S. Shahbazpanahi, T. Kirubarajan
{"title":"Performance analysis of blind adaptive MIMO receivers","authors":"B. Balasingam, M. Bolic, S. Shahbazpanahi, T. Kirubarajan","doi":"10.1109/ICASSP.2010.5495966","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495966","url":null,"abstract":"In this paper, we derive a theoretical performance evaluation scheme of Kalman filter based channel tracking and data decoding for multiple-input multiple-output orthogonal frequency division multiplexed (MIMO-OFDM) communication systems that are based on orthogonal space-time block codes. The derivation is approximate, however, it is novel and demonstrated accurate for practical scenarios. Assuming a prior distribution for the initial channel we have derived the instantaneous signal to interference and noise ratio (SINR) for consecutive transmission blocks in the absence of training by exploiting Kalman filtering to track the channel. A theoretical estimation of BER is then derived based on such instantaneous SINR values. The resulting analysis is able to study the effect of different parameters of the system such as the number of antennas, number of sub-carriers, mobile velocity and the assumed channel length on the BER performance of the system. Numerical examples confirm the validity of the theoretical analysis.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121196910","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}
引用次数: 4
Fishervioce: A discriminant subspace framework for speaker recognition fisher voice:说话人识别的判别子空间框架
2010 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2010-03-14 DOI: 10.1109/ICASSP.2010.5495591
Zhifeng Li, W. Jiang, H. Meng
{"title":"Fishervioce: A discriminant subspace framework for speaker recognition","authors":"Zhifeng Li, W. Jiang, H. Meng","doi":"10.1109/ICASSP.2010.5495591","DOIUrl":"https://doi.org/10.1109/ICASSP.2010.5495591","url":null,"abstract":"We propose a new framework for speaker recognition, referred as Fishervoice. It includes the design of a feature representation known as the structured score vector (SSV), which relates acoustic structures with “key” frames in an input utterance in capturing relevant speaker characteristics. The framework also applies nonparametric Fisher's discriminant analysis to map the SSVs into a compressed discriminant subspace, where matching is performed between a test sample and reference speaker samples to achieve speaker recognition. The objective is to reduce intra-speaker variability and emphasize discriminative class boundary information to facilitate speaker recognition. Experiments based on the XM2VTSDB corpus shows that the Fishervoice framework gave superior performance, compared with other commonly used approaches, e.g. GMM-UBM and Eigenvoice.","PeriodicalId":293333,"journal":{"name":"2010 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"209 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2010-03-14","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121203842","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}
引用次数: 13
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