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Algebraic visual cryptography scheme for color images
, Article 2008 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP, Las Vegas, NV, 31 March 2008 through 4 April 2008 ; 2008 , Pages 1761-1764 ; 15206149 (ISSN) ; 1424414849 (ISBN); 9781424414840 (ISBN) ; Alamdar Yazdi, A ; Plataniotis, K. N ; Sharif University of Technology
2008
Abstract
This paper introduces a novel, cost effective visual cryptography scheme suitable for color image transmission over bandwidth constraint channels. Unlike previously proposed schemes, the solution offers perfect reconstruction while producing shares with size smaller than that of the input image. The maximum distance separable (MDS) code principle used in the design allows for the introduction of a flexible framework that compares favorably to competing solutions as it can be seen by examining the experimental results included in this paper. ©2008 IEEE
Complex-valued sparse representation based on smoothed ℓ0 norm
, Article 2008 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP, Las Vegas, NV, 31 March 2008 through 4 April 2008 ; 2008 , Pages 3881-3884 ; 15206149 (ISSN) ; 1424414849 (ISBN); 9781424414840 (ISBN) ; Babaie Zadeh, M ; Jutten, C ; Sharif University of Technology
2008
Abstract
In this paper we present an algorithm for complex-valued sparse representation. In our previous work we presented an algorithm for Sparse representation based on smoothed l0-norm. Here we extend that algorithm to complex-valued signals. The proposed algorithm is compared to FOCUSS algorithm and it is experimentally shown that the proposed algorithm is about two or three orders of magnitude faster than FOCUSS while providing approximately the same accuracy. ©2008 IEEE
A first step to convolutive sparse representation
, Article 2008 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP, Las Vegas, NV, 31 March 2008 through 4 April 2008 ; 2008 , Pages 1921-1924 ; 15206149 (ISSN) ; 1424414849 (ISBN); 9781424414840 (ISBN) ; Babaie Zadeh, M ; Ghasemian Sahebi, A ; Jutten, C ; Sharif University of Technology
2008
Abstract
In this paper an extension of the sparse decomposition problem is considered and an algorithm for solving it is presented. In this extension, it is known that one of the shifted versions of a signal s (not necessarily the original signal itself) has a sparse representation on an overcomplete dictionary, and we are looking for the sparsest representation among the representations of all the shifted versions of s. Then, the proposed algorithm finds simultaneously the amount of the required shift, and the sparse representation. Experimental results emphasize on the performance of our algorithm. ©2008 IEEE
A high-performance architecture for irregular LDPC decoding algorithm using input-multiplexing method
, Article 2007 9th International Symposium on Signal Processing and its Applications, ISSPA 2007, Sharjah, 12 February 2007 through 15 February 2007 ; 2007 ; 1424407796 (ISBN); 9781424407798 (ISBN) ; Mohtashami, V ; Sharif University of Technology
2007
Abstract
A new high-performance architecture for decoding the irregular Low-Density Parity-Check (LDPC) codes with respect to the iterative message-passing decoding algorithm is explored. The proposed method is based on reducing the logic delays in the iterative processing of the bit nodes and check nodes leading to the increment of maximum possible frequency. The simulations show the efficiency of the proposed method in low/high-complexity graph matrices, though it is more effective in high-complexity ones. About 28% reduction of the combinational delay in the bit/check processors is explored without much impacting the area consumption. ©2007 IEEE
Secure consensus averaging for secure information fusion in sensor networks
, Article 2007 9th International Symposium on Signal Processing and its Applications, ISSPA 2007, Sharjah, 12 February 2007 through 15 February 2007 ; 2007 ; 1424407796 (ISBN); 9781424407798 (ISBN) ; Talebi, M. S ; Rabiee, H. R ; Khalaj, B. H ; Sharif University of Technology
2007
Abstract
In this work, we have examined the problem of distributed consensus averaging over senor networks from a novel point of view considering the need for security. We have proposed a method for incorporating privacy into the scalable average consensus mechanisms. Our proposed method, Random Projections Method (RPM), is lightweight and transparent since it is not based on cryptography and does not require any change in the fusion system. RPM is based on introducing a simple, yet effective pre-fusion algorithm. We mathematically derived the correctness of RPM and analyzed its effect on convergence of the system through simulation. Robustness of RPM against honest-but-curious adversaries is...
Effects of physical exercise on the photoplethysmogram waveform
, Article 2007 5th Student Conference on Research and Development, SCORED, Selangor, 11 December 2007 through 12 December 2007 ; 2007 ; 1424414709 (ISBN); 9781424414703 (ISBN) ; Zahedi, E ; Mohd Ali, M. A ; Sharif University of Technology
2007
Abstract
Photoplethysmography is a non-invasive method which is suitable to estimate vascular compliance. Photoplethysmogram (PPG) waveform has been used to evaluate vascular characteristic changes due to exercise. Quantification of the primary and secondary peak position is of interest as a potential objective, individualized measure of the level of exercise that a subject has achieved with respect to a baseline status, prior to the beginning of the exercise session. The modified bicycle ergonomic protocol was selected for this experiment. Two different signal processing methods were employed in processing the PPG waveform to characterize its changes through a single parameter. Both, the visual...
Design and implementation of a PC-based digital synchronous detection system for biological signal measurement
, Article 2007 5th Student Conference on Research and Development, SCORED, Selangor, 11 December 2007 through 12 December 2007 ; 2007 ; 1424414709 (ISBN); 9781424414703 (ISBN) ; Zahedi, E ; Mohd Ali, M. A ; Sharif University of Technology
2007
Abstract
A PC-based digital lock-in amplifier was implemented completely in software (Labview) for photoplethysmograph (PPG) measurement. Experiments carried out to evaluate the performance of the amplifier show that the system can recover the signals up to 40 dB below the interference (ambient light). This system can be customized later to measure the trans-abdominal PPG signal of pregnant women for fetal heart rate detection. ©2007 IEEE
ECG baseline correction with adaptive bionic wavelet transform
, Article 2007 9th International Symposium on Signal Processing and its Applications, ISSPA 2007, Sharjah, 12 February 2007 through 15 February 2007 ; 2007 ; 1424407796 (ISBN); 9781424407798 (ISBN) ; Shamsollahi, M. B ; Sharif University of Technology
2007
Abstract
We have presented a new method for ECG baseline correction using the adaptive bionic wavelet transform (BWT). In fact by the means of BWT, the resolution in the time-frequency domain can be adaptively adjusted not only by the signal frequency but also by the signal instantaneous amplitude and its first-order differential. Besides by optimizing the BWT parameters parallel to modifying our previous thresholding rule, one can handle ECG baseline correction. First an estimation of the baseline wandering frequency is obtained and then the adaptation can be used only in three successive scales in which the mid-scale has the closest center frequency to the estimated frequency. Thus the...
Noise and speaker robustness in a persian continuous speech recognition system
, Article 2007 9th International Symposium on Signal Processing and its Applications, ISSPA 2007, Sharjah, 12 February 2007 through 15 February 2007 ; 2007 ; 1424407796 (ISBN); 9781424407798 (ISBN) ; Sameti, H ; Sharif University of Technology
2007
Abstract
In this paper VTLN speaker normalization, MLLR and MAP adaptation methods are investigated in a Persian HMM-based speaker independent large vocabulary continuous speech recognition system. Speaker and environmental noise robustness are achieved in real world applications for this system. A search-based method is used in VTLN to find speaker relative warping factors. The warping factors are applied to signal's spectrum to normalize the variation effect of VTL between speakers. In the MLLR framework, Gaussian mean and covariance transformations in global and full adaptation are experienced. In this method, regression tree based adaptation in batch-supervised fashion is used. Also the standard...
PIRS: Pseudo inversion based recovery of speech signals
, Article ISSPIT 2007 - 2007 IEEE International Symposium on Signal Processing and Information Technology, Cairo, 15 December 2007 through 18 December 2007 ; 2007 , Pages 285-290 ; 9781424418350 (ISBN) ; Lakdashti, A ; Manzuri Shalmani, M. T ; Sharif University of Technology
2007
Abstract
Communication of speech over error prone channels such as wireless channels and internet usually suffers from loss of large number of adjacent samples. In this paper, we propose to make artificial correlation between speech samples which distorts it. By choosing appropriate parameters, one can control this distortion to lie below acceptable ranges. Using this correlation, the receiver can recover lost samples up to a certain limit using our proposed algorithm. Experimental results show that our solution overcomes a previous one reported in the literature specially when the amount of lost samples are below the mentioned limit. ©2007 IEEE
A new high-speed class-AB current-mode circuit
, Article 2007 IEEE International Symposium on Circuits and Systems, ISCAS 2007, New Orleans, LA, 27 May 2007 through 30 May 2007 ; 2007 , Pages 717-720 ; 02714310 (ISSN) ; Bakhtiar, M. S ; Sharif University of Technology
Institute of Electrical and Electronics Engineers Inc
2007
Abstract
this paper presents a new class AB circuit for current-mode signal processing. The proposed circuit provides high-dynamic range, low distortion and accurate definition of quiescent current and it is well suited for high-speed applications. A third-order low-pass filter with a cutoff frequency of 200MHz and 51dB dynamic range is also presented. The Alter consumes 2.7mW from 1.8V supply. © 2007 IEEE
On a decentralized deterministic transmission power level selection algorithm in large aloha networks under saturation
, Article 2006 IEEE International Conference on Communications, ICC 2006, Istanbul, 11 July 2006 through 15 July 2006 ; Volume 12 , 2006 , Pages 5732-5737 ; 05361486 (ISSN); 1424403553 (ISBN); 9781424403554 (ISBN) ; Khalaj, B. H ; Sharif University of Technology
2006
Abstract
In this paper, we will analyze a decentralized transmission power level selection algorithm in conjunction with Binary Exponential Back-off retransmission scheme, in large slotted-Aloha networks under saturation condition. In this algorithm, the transmission power is steadily increased by an amount called power step, until the packet is successfully transmitted. We will use SIR-based capture model to study the performance of this algorithm with small and large power steps. Our analysis proves that, although small power steps may seem favorable for power saving purposes, these power steps face throughput collapse under high load conditions (large number of nodes), while properly chosen large...
A new method for estimating Score Function Difference (SFD) and its application to Blind Source Separation
, Article 13th European Signal Processing Conference, EUSIPCO 2005, Antalya, 4 September 2005 through 8 September 2005 ; 2005 , Pages 1507-1510 ; 1604238216 (ISBN); 9781604238211 (ISBN) ; Babaie Zadeh, M ; Jutten, C ; Sharif University of Technology
2005
Abstract
Score Function Difference (SFD) is a recently proposed "gradient" for mutual information which can be used in Blind Source Separation algorithms based on minimization of mutual information. To be applied to practical problems, SFD must be estimated from the data samples. In this paper, a new method for estimating SFD is proposed. To compare the performance of this new estimator with other proposed SFD estimation methods, we have applied them in separating linear instantaneous mixtures. It will be seen that our method performs superior to all other methods previously proposed for estimation of SFD
Measuring minimum critical flow for normal breath sounds
, Article 2005 27th Annual International Conference of the Engineering in Medicine and Biology Society, IEEE-EMBS 2005, Shanghai, 1 September 2005 through 4 September 2005 ; Volume 7 VOLS , 2005 , Pages 2726-2729 ; 05891019 (ISSN); 0780387406 (ISBN); 9780780387409 (ISBN) ; Moussavi, Z ; Sharif University of Technology
Institute of Electrical and Electronics Engineers Inc
2005
Abstract
Relationship between respiratory sounds and flow has always been of interest for researchers and physicians. However, the flow-sound relationship at very low flow rate has been questionable because breath sounds must exceed a minimum flow in order to be audible and different from the background noise. This study aimed to find the minimum critical flow rates for respiratory sounds to be audible and different from background noise. Tracheal and lung sound signals of healthy subjects in two groups of adults (12 subjects) and children (9 subjects) were studied. The values of minimum critical flow were determined comparing the spectrogram of the respiratory sounds at very low flow with that of...
CIROLS: Codec independent recovery of lost speech packets
, Article 2007 9th International Symposium on Signal Processing and its Applications, ISSPA 2007, Sharjah, 12 February 2007 through 15 February 2007 ; 2007 ; 1424407796 (ISBN); 9781424407798 (ISBN) ; Manzuri Shalmani, M. T ; Aghatabar, M. M ; Sharif University of Technology
2007
Abstract
In this paper, we have focused on finding an error resilient method for discontinuity-less transmission of speech signals in the internet. Our proposed method creates artificial correlation between speech samples that pre-distorts the speech signal. The receiver uses this correlation to reconstruct the lost speech packets. A discrete Fourier transform (DFT)-based speech enhancement technique is designed for the reduction of the processing error in the recovered speech caused by the speech codecs. The SegSNR results show the superiority of our proposed method over a recently proposed speech enhancement technique. ©2007 IEEE
A new pipeline implementation of an adaptive IIR filter for noise reduction application
, Article IEEE International Symposium on Communications and Information Technologies: Smart Info-Media Systems, ISCIT 2004, Sapporo, 26 October 2004 through 29 October 2004 ; Volume 1 , 2004 , Pages 577-581 ; 0780385934 (ISBN) ; Manzuri, M. T ; Ayat, S ; Sharif University of Technology
2004
Abstract
the parallel form in adaptive HR filtering is an efficient realization that provides robust stability monitoring with less complexity than that of the direct form. This paper presents an implementation of two line parallel structure employing a real orthogonal transform and real coefficients second-order section as sub filters on FPGA. Architecture that is implemented is in pipeline fashion and it is independent from its building blocks and they may have its own implementation or in other system numbering. In the last section, we compared throughput of our architecture with a Ti DSP (TMS320C3x/4x), the results show that this architecture works better than the DSP
A Minimization-Projection (MP) approach for blind separating convolutive mixtures
, Article Proceedings - IEEE International Conference on Acoustics, Speech, and Signal Processing, Montreal, Que, 17 May 2004 through 21 May 2004 ; Volume 5 , 2004 , Pages V-533-V-536 ; 15206149 (ISSN) ; Jutten, C ; Nayebi, K ; Sharif University of Technology
2004
Abstract
In this paper, a new algorithm for blind source separation in convolutive mixtures, based on minimizing the mutual information of the outputs, is proposed. This minimization is done using a recently proposed Minimization-Projection (MP) approach for minimizing mutual information in a parametric model. Since the minimization step of the MP approach is proved to have no local minimum, it is expected that this new algorithm has good convergence behaviours
A novel two frequency MTI radar
, Article Proceedings of the IEEE Radar Conference, Philadelphia, PA, 26 April 2004 through 29 April 2004 ; 2004 , Pages 589-591 ; Norouzi, Y ; Nayebi, M. M ; Sharif University of Technology
2004
Abstract
The paper introduced a new design for two-frequency MTI radar is introduced. The suggested system can change its frequencies, in each pulse. Therefore, the system is very resistive to electronic war. The analytical results of our calculation show that the system has very high blind speed and in realistic situations, it increases signal to noise ratio, although it widens clutter bandwidth and detects some spurious targets
Feedforward multiple-input active noise control systems
, Article 2003 ASME International Mechanical Engineering Congress, Washington, DC., 15 November 2003 through 21 November 2003 ; Volume 72, Issue 1 , 2003 , Pages 143-150 ; Mehdigholl, H ; Esmailzadeh, E ; Sharif University of Technology
American Society of Mechanical Engineers (ASME)
2003
Abstract
The use of adaptive feedforward controllers has proven to be a very successful strategy for controlling noise and vibration in a variety of applications. One reason is that the feedforward controller is an open loop controller, which can be designed to cancel the undesired noise in one position with any accuracy. However, the feedforward controller requires an input signal, called a reference signal, correlated to the noise source. As a consequence, a single reference controller can only reduce noise radiated from a single noise source. In many applications, there is a need to attenuate noise produced by several noise sources. In this paper, three different structures, single, modulating and...
Composite pnlms & nlms adaptation: A new method for network echo cancellation
, Article 14th International Conference on Digital Signal Processing, DSP 2002, 1 July 2002 through 3 July 2002 ; Volume 2 , 2002 , Pages 757-760 ; 0780375033 (ISBN) ; Atarodi, M ; Sharif University of Technology
Institute of Electrical and Electronics Engineers Inc
2002
Abstract
This paper describes an improved version of the recently proposed fast converging algorithm. PNLMS, for network echo cancellers. We have introduced a simple analysis of the PNLMS convergence behavior to show why after the fast initial convergence, it slows down. Also, the method has been worked out to overcome this deficiency is presented. Improvement is shown by several simulation results. © 2002 IEEE