Pub Date : 2011-03-24DOI: 10.1109/ICCSP.2011.5739327
P. Beinschob, U. Zolzer
OFDM-based multi antenna systems suffer from inaccurate channel estimates in time variant channels which lead eventually to performance degradation. Decision-directed channel estimation (DDCE) schemes have been proposed to reduce the number of reference symbols. A prediction of channel coefficients is necessary for MIMO detection, whose output is used to acquire new channel estimates recursively. Three approaches to predict channel coefficients are investigated in this work. For validation simulation were conducted with the 3GPP spatial channel model. Channel estimation and prediction accuracy for mobile terminals in a range of velocities and Signal to Noise Ratios were evaluated. Even though the predictors vary strong in computational complexity it is shown in this work the performance benefit of complicated approaches remains small in a variety of channel states.
{"title":"Predictor performance of decision-directed channel estimation in 3GPP MIMO channels","authors":"P. Beinschob, U. Zolzer","doi":"10.1109/ICCSP.2011.5739327","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739327","url":null,"abstract":"OFDM-based multi antenna systems suffer from inaccurate channel estimates in time variant channels which lead eventually to performance degradation. Decision-directed channel estimation (DDCE) schemes have been proposed to reduce the number of reference symbols. A prediction of channel coefficients is necessary for MIMO detection, whose output is used to acquire new channel estimates recursively. Three approaches to predict channel coefficients are investigated in this work. For validation simulation were conducted with the 3GPP spatial channel model. Channel estimation and prediction accuracy for mobile terminals in a range of velocities and Signal to Noise Ratios were evaluated. Even though the predictors vary strong in computational complexity it is shown in this work the performance benefit of complicated approaches remains small in a variety of channel states.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"170 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124244882","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739304
J. Jena, T. Kishore Kumar
The aim of this paper is development and implementation of a unconventional method to determine the position of artillery shells using acoustic signature. Hence the authors developed an algorithm to achieve the same. Analysis has been done to determine the error introduced by the various sources on the final position determined by the algorithm.
{"title":"Determination of position of detonation of artillery shells","authors":"J. Jena, T. Kishore Kumar","doi":"10.1109/ICCSP.2011.5739304","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739304","url":null,"abstract":"The aim of this paper is development and implementation of a unconventional method to determine the position of artillery shells using acoustic signature. Hence the authors developed an algorithm to achieve the same. Analysis has been done to determine the error introduced by the various sources on the final position determined by the algorithm.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122825345","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739310
Sajan Goud, M. Jacob
We recently proposed an accelerated dynamic magnetic resonance imaging (MRI) reconstruction algorithm that exploits the underlying low rank and sparse properties of the data to achieve highly accelerated reconstructions. In this paper, we validate our algorithm in the context of dynamic free breathing cardiac Perfusion MRI on the Physiologically Improved Non Uniform Cardiac Torso Phantom, PINCAT phantom. The practical utilities of our scheme in providing significantly better reconstructions at higher accelerations in comparison to existing methods are studied. We demonstrate that our scheme do not have trade offs with accurate temporal modeling and spatial quality unlike the existing low rank based schemes. Our results also show the capability of our scheme to achieve better reconstruction qualities at high accelerations in comparison to using only the low rank or sparsity properties individually. We argue that the speed up obtained by our scheme could be capitalized in perfusion imaging to provide better spatio-temporal resolutions and volume coverage while the subject is freely breathing.
{"title":"Free breathing cardiac perfusion MRI reconstruction using a sparse and low rank model: Validation with the Physiologically Improved NCAT phantom","authors":"Sajan Goud, M. Jacob","doi":"10.1109/ICCSP.2011.5739310","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739310","url":null,"abstract":"We recently proposed an accelerated dynamic magnetic resonance imaging (MRI) reconstruction algorithm that exploits the underlying low rank and sparse properties of the data to achieve highly accelerated reconstructions. In this paper, we validate our algorithm in the context of dynamic free breathing cardiac Perfusion MRI on the Physiologically Improved Non Uniform Cardiac Torso Phantom, PINCAT phantom. The practical utilities of our scheme in providing significantly better reconstructions at higher accelerations in comparison to existing methods are studied. We demonstrate that our scheme do not have trade offs with accurate temporal modeling and spatial quality unlike the existing low rank based schemes. Our results also show the capability of our scheme to achieve better reconstruction qualities at high accelerations in comparison to using only the low rank or sparsity properties individually. We argue that the speed up obtained by our scheme could be capitalized in perfusion imaging to provide better spatio-temporal resolutions and volume coverage while the subject is freely breathing.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"32 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134043737","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739348
D. Murali Mohan, Dileep B. Karpur, M. Narayan, J. Kishore
Spectrum of speech signals have frequency components from 50Hz to 7 kHz (Wideband speech). However, due to historical reasons speech is band-pass filtered between 300 Hz-3.4 kHz in PSTN networks and this speech is referred to as narrowband speech. The missing bandwidth in narrow band speech contributes to speech quality and intelligibility. This paper addresses the problem of artificial bandwidth extension of narrowband speech to wideband speech. The proposed method for bandwidth extension is based on statistical recovery using Gaussian Mixture Model (GMM) for spectral envelope parameters and spectral shifting method is used for excitation extension.
{"title":"Artificial bandwidth extension of narrowband speech using Gaussian Mixture Model","authors":"D. Murali Mohan, Dileep B. Karpur, M. Narayan, J. Kishore","doi":"10.1109/ICCSP.2011.5739348","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739348","url":null,"abstract":"Spectrum of speech signals have frequency components from 50Hz to 7 kHz (Wideband speech). However, due to historical reasons speech is band-pass filtered between 300 Hz-3.4 kHz in PSTN networks and this speech is referred to as narrowband speech. The missing bandwidth in narrow band speech contributes to speech quality and intelligibility. This paper addresses the problem of artificial bandwidth extension of narrowband speech to wideband speech. The proposed method for bandwidth extension is based on statistical recovery using Gaussian Mixture Model (GMM) for spectral envelope parameters and spectral shifting method is used for excitation extension.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134088620","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739315
Vivek Dhoot, Sanjeev Gupta
In this paper, a novel multifractal cantor based multiband monopole antenna is proposed and analyzed using 3-Dimensional Finite Difference Time Domain Method (3D-FDTD). The proposed antenna has multiband characteristics covering several wireless applications in Ultra Wideband (UWB) including WLAN 2.4 GHz and 5.8 GHz, GSM, PCS and DCS applications. A program based on 3D-FDTD method is written and utilized for observing return loss of the proposed antenna.
{"title":"Full wave analysis of a novel multifractal multiband antenna using 3D-FDTD approach","authors":"Vivek Dhoot, Sanjeev Gupta","doi":"10.1109/ICCSP.2011.5739315","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739315","url":null,"abstract":"In this paper, a novel multifractal cantor based multiband monopole antenna is proposed and analyzed using 3-Dimensional Finite Difference Time Domain Method (3D-FDTD). The proposed antenna has multiband characteristics covering several wireless applications in Ultra Wideband (UWB) including WLAN 2.4 GHz and 5.8 GHz, GSM, PCS and DCS applications. A program based on 3D-FDTD method is written and utilized for observing return loss of the proposed antenna.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"50 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132534471","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739352
Paramjeet Singh, A. K. Verma
The combined Quasi Static Spectral Domain Approach (SDA) method and Single Layer Reduction (SLR) technique is presented to compute dielectric loss of multilayer Coplanar Waveguide (CPW). The Green's function for the multilayer structure is derived from Transverse Transmission Line (TTL) method. Quasi static SDA method is used to compute effective relative permittivity of the multilayer CPW. The Single Layer Reduction (SLR) technique converts multilayer CPW structure to an equivalent single layer CPW structure. The dielectric loss is computed for the equivalent CPW structure.
{"title":"Dielectric loss computation of multilayer Coplanar Waveguide","authors":"Paramjeet Singh, A. K. Verma","doi":"10.1109/ICCSP.2011.5739352","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739352","url":null,"abstract":"The combined Quasi Static Spectral Domain Approach (SDA) method and Single Layer Reduction (SLR) technique is presented to compute dielectric loss of multilayer Coplanar Waveguide (CPW). The Green's function for the multilayer structure is derived from Transverse Transmission Line (TTL) method. Quasi static SDA method is used to compute effective relative permittivity of the multilayer CPW. The Single Layer Reduction (SLR) technique converts multilayer CPW structure to an equivalent single layer CPW structure. The dielectric loss is computed for the equivalent CPW structure.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132668601","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739371
Sunil P. Joshi, R. Paily
The use of error-correcting codes has proven to be an effective way to overcome data corruption in digital wireless communication channels, enabling reliable transmission over noisy and fading channel. This requires low power decoders as they consume lot of power. Power reduction in any system can be achieved at device level, at circuit level or at architectural level. In this paper, power reduction is achieved at architecture level. A Viterbi Decoder (VD) with architectural modification for Add-Compare-Select Unit (ACSU) and clock gated Survivor Memory Unit (SMU) are designed for low power wireless applications. A decoder system with code rate of k/n=1/2 with constraint length K=7 has been implemented with 130nm technology. It is synthesized using design compiler of Synopsys and its power is estimated with power compiler. A throughput of 125 Mbps is achieved satisfying the requirement for wireless applications. Bit error rate of proposed system is same as that of modified register exchange VD. Around 66% power is reduced with clock gating technique.
{"title":"Low power Viterbi Decoder by modified ACSU architecture and clock gating method","authors":"Sunil P. Joshi, R. Paily","doi":"10.1109/ICCSP.2011.5739371","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739371","url":null,"abstract":"The use of error-correcting codes has proven to be an effective way to overcome data corruption in digital wireless communication channels, enabling reliable transmission over noisy and fading channel. This requires low power decoders as they consume lot of power. Power reduction in any system can be achieved at device level, at circuit level or at architectural level. In this paper, power reduction is achieved at architecture level. A Viterbi Decoder (VD) with architectural modification for Add-Compare-Select Unit (ACSU) and clock gated Survivor Memory Unit (SMU) are designed for low power wireless applications. A decoder system with code rate of k/n=1/2 with constraint length K=7 has been implemented with 130nm technology. It is synthesized using design compiler of Synopsys and its power is estimated with power compiler. A throughput of 125 Mbps is achieved satisfying the requirement for wireless applications. Bit error rate of proposed system is same as that of modified register exchange VD. Around 66% power is reduced with clock gating technique.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"27 3","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132900938","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739379
S. Siddamal, R. Banakar, B. C. Jinaga
This paper describes the architecture of system level design for the analysis of fiber parameters for one simulation step considering the synchronous and timing issues. The challenge in realizing these systems is not only the hardware but also complex control design that marshals the data flow. In a well-thought-out system level design approach it is necessary in splitting the design into several sub-modules, each addressing the specific timing and synchronizing issues. For the split step Fourier algorithm a system level model is designed considering the data path and control architecture. The timing and synchronizing are considering in RTL validation using Xilinx device XC5VLX30TFF655 with speed grade −3.
{"title":"Timing consideration in synchronous system level design","authors":"S. Siddamal, R. Banakar, B. C. Jinaga","doi":"10.1109/ICCSP.2011.5739379","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739379","url":null,"abstract":"This paper describes the architecture of system level design for the analysis of fiber parameters for one simulation step considering the synchronous and timing issues. The challenge in realizing these systems is not only the hardware but also complex control design that marshals the data flow. In a well-thought-out system level design approach it is necessary in splitting the design into several sub-modules, each addressing the specific timing and synchronizing issues. For the split step Fourier algorithm a system level model is designed considering the data path and control architecture. The timing and synchronizing are considering in RTL validation using Xilinx device XC5VLX30TFF655 with speed grade −3.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127691207","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739300
A. Revathi, Y. Venkataramani
The main objective of this paper is to explore the effectiveness of perceptual features for performing isolated digits and continuous speech recognition. The proposed perceptual features are captured and code book indices are extracted. Expectation maximization algorithm is used to generate HMM models for the speeches. Speech recognition system is evaluated on clean test speeches and the experimental results reveal the performance of the proposed algorithm in recognizing isolated digits and continuous speeches based on maximum log likelihood value between test features and HMM models for each speech. Performance of these features is tested on speeches randomly chosen from “TI Digits_1”, “TI Digits_2” and “TIMIT” databases. This algorithm is tested for VQ and combination of VQ and HMM speech modeling techniques. Perceptual linear predictive cepstrum yields the accuracy of 86% and 93% for speaker independent isolated digit recognition using VQ and combination of VQ & HMM speech models respectively. This feature also gives 99% and 100% accuracy for speaker independent continuous speech recognition by using VQ and the combination of VQ & HMM speech modeling techniques.
{"title":"Speaker independent continuous speech and isolated digit recognition using VQ and HMM","authors":"A. Revathi, Y. Venkataramani","doi":"10.1109/ICCSP.2011.5739300","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739300","url":null,"abstract":"The main objective of this paper is to explore the effectiveness of perceptual features for performing isolated digits and continuous speech recognition. The proposed perceptual features are captured and code book indices are extracted. Expectation maximization algorithm is used to generate HMM models for the speeches. Speech recognition system is evaluated on clean test speeches and the experimental results reveal the performance of the proposed algorithm in recognizing isolated digits and continuous speeches based on maximum log likelihood value between test features and HMM models for each speech. Performance of these features is tested on speeches randomly chosen from “TI Digits_1”, “TI Digits_2” and “TIMIT” databases. This algorithm is tested for VQ and combination of VQ and HMM speech modeling techniques. Perceptual linear predictive cepstrum yields the accuracy of 86% and 93% for speaker independent isolated digit recognition using VQ and combination of VQ & HMM speech models respectively. This feature also gives 99% and 100% accuracy for speaker independent continuous speech recognition by using VQ and the combination of VQ & HMM speech modeling techniques.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"63 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116278919","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 : 2011-03-24DOI: 10.1109/ICCSP.2011.5739361
P. Shanmugapriya, Y. Venkataramani
A Fuzzy Wavelet network (FWN) is proposed to model the characteristics of a speaker in an automatic speaker verification system in this paper. The neural network using wavelet as activation function is wavelet network (Wavenet). Wavenet has the ability to extract the distinguishable and essential features in frequency rich signals. This is required in classification and identification problems such as speaker verification. Nonlinearity and structured knowledge representation with human perception of fuzzy inference system makes it to be a suitable model for speaker verification when combined with the wavelet network. In this approach, the wavelet theory is combined with the fuzzy based neural network theory which leads to construction of Fuzzy Wavelet Network (FWN). The advantage of fuzzy wavelet network is that the membership functions can be easily merged or divided using the multi resolution properties and the rules can be evaluated during learning. The performance of the proposed speaker verification system is evaluated with TIMIT database. A comparison is made between the proposed system and the system using state of the art model (GMM). Compared with GMM and WNN, FWN provides better verification performance.
{"title":"Implementation of speaker verification system using Fuzzy Wavelet Network","authors":"P. Shanmugapriya, Y. Venkataramani","doi":"10.1109/ICCSP.2011.5739361","DOIUrl":"https://doi.org/10.1109/ICCSP.2011.5739361","url":null,"abstract":"A Fuzzy Wavelet network (FWN) is proposed to model the characteristics of a speaker in an automatic speaker verification system in this paper. The neural network using wavelet as activation function is wavelet network (Wavenet). Wavenet has the ability to extract the distinguishable and essential features in frequency rich signals. This is required in classification and identification problems such as speaker verification. Nonlinearity and structured knowledge representation with human perception of fuzzy inference system makes it to be a suitable model for speaker verification when combined with the wavelet network. In this approach, the wavelet theory is combined with the fuzzy based neural network theory which leads to construction of Fuzzy Wavelet Network (FWN). The advantage of fuzzy wavelet network is that the membership functions can be easily merged or divided using the multi resolution properties and the rules can be evaluated during learning. The performance of the proposed speaker verification system is evaluated with TIMIT database. A comparison is made between the proposed system and the system using state of the art model (GMM). Compared with GMM and WNN, FWN provides better verification performance.","PeriodicalId":408736,"journal":{"name":"2011 International Conference on Communications and Signal Processing","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-03-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125189890","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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