Search for: digital-signal-processing
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    Digesting network traffic for forensic investigation using digital signal processing techniques

    , Article IEEE Transactions on Information Forensics and Security ; Volume 14, Issue 12 , 2019 , Pages 3312-3321 ; 15566013 (ISSN) Mohammad Hosseini, S ; Jahangir, A.H ; Kazemi, M ; Sharif University of Technology
    Institute of Electrical and Electronics Engineers Inc  2019
    One of the most important practices of cybercrime investigations is to search a network traffic history for an excerpt of traffic, such as the disclosed information of an organization or a worm's signature. In post-mortem investigations, criminals and targets are detected by attributing the excerpt to payloads of traffic flows. Since it is impossible to store the high volume of data related to long-term traffic history, payload attribution systems (PASs) based on storing a compact digest of traffic using Bloom filters have been presented in the literature. In these systems, querying the stored digest for an excerpt results in a low number of suspects instead of certain criminals. In this... 

    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) Golmohammadi, A ; Manzuri, M. T ; Ayat, S ; Sharif University of Technology
    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  

    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) Nekuii, M ; Atarodi, M ; Sharif University of Technology
    Institute of Electrical and Electronics Engineers Inc  2002
    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  

    An image annotation rectifying method based on deep features

    , Article 2nd International Conference on Digital Signal Processing, ICDSP 2018, 25 February 2018 through 27 February 2018 ; 2018 , Pages 88-92 ; 9781450364027 (ISBN) Ghostan Khatchatoorian, A ; Jamzad, M ; Sharif University of Technology
    Association for Computing Machinery  2018
    Automatic image annotation methods generate a list of tags for each test image and present it in a matrix structure. To achieve a more accurate annotation, we propose a method with the aim of correcting the tag list. In our method, we detect an indicator for each group of tags and use it to rectify the annotation results. To find a correct indicator, we apply a deep feature vector generated by the “AlexNet” model. Using this indicator, we determine the suitable tags for an image. The purposed method is independent of feature vector, dataset, and annotation method. It can be applied to the currently available annotation methods. Our experiments showed improvement in all annotation methods... 

    Two channel abdominal PPG instrumentation

    , Article IFMBE Proceedings ; Volume 21 IFMBE, Issue 1 , 2008 , Pages 691-693 ; 16800737 (ISSN); 9783540691389 (ISBN) Gan, K. B ; Mohd Ali, M. A ; Zahedi, E ; Sharif University of Technology
    Springer Verlag  2008
    In this paper, a two channel abdominal PPG instrumentation is developed. The proposed instrument consists of IR-LED and its driver, photo-detector and data acquisition card. The modulation frequency generation, demodulation and digital signal processing is done completely in the digital domain using LabView. The results show that the developed instrument is able to acquire signal from the abdomen even at 4 cm source to detector separation. This instrument is intended for future application in trans-abdominal fetal heart rate detection. © 2008 Springer-Verlag  

    The edge product of networks

    , Article 18th International Conference on Parallel and Distributed Computing, Applications and Technologies, PDCAT 2007, Adelaide, SA, 3 December 2007 through 6 December 2007 ; January , 2007 , Pages 371-375 ; 0769530494 (ISBN); 9780769530499 (ISBN) Jalali, A ; Sarbazi Azad, H ; Sharif University of Technology
    In this paper, a new graph product, called Edge Graph Product (EGP) is proposed by replacing each edge in the multiplicand graph by a copy of the multiplier graph via two candidate nodes. The edge product, unlike other products already proposed, results in a graph whose number of edges is numerical product of the number of the edges in the multiplicand and multiplier graphs, and the number of vertices is not equal to the numerical product of the number of vertices in the multiplicand and multiplier graphs. After formal definition of the new product, some basic properties of the product operator are studied. We then address Hamiltonian, Eulerian and routing properties of the new product, and... 

    Source estimation in noisy sparse component analysis

    , Article 2007 15th International Conference onDigital Signal Processing, DSP 2007, Wales, 1 July 2007 through 4 July 2007 ; July , 2007 , Pages 219-222 ; 1424408822 (ISBN); 9781424408825 (ISBN) Zayyani, H ; Babaiezadeh, M ; Jutten, C ; Sharif University of Technology
    In this paper, a new algorithm for Sparse Component Analysis (SCA) in the noisy underdetermined case (i.e., with more sources than sensors) is presented. The solution obtained by the proposed algorithm is compared to the minimum l1 -norm solution achieved by Linear Programming (LP). Simulation results show that the proposed algorithm is approximately 10 dB better than the LP method with respect to the quality of the estimated sources. It is due to optimality of our solution (in the MAP sense) for source recovery in noisy underdetermined sparse component analysis in the case of spiky model for sparse sources and Gaussian noise. © 2007 IEEE  

    Traffic improvements in wireless communication networks using antenna arrays

    , Article IEEE Journal on Selected Areas in Communications, Piscataway, NJ, United States ; Volume 18, Issue 3 , 2000 , Pages 458-471 ; 07338716 (ISSN) Razavilar, J ; Rashid Farrokhi, F ; Liu, K. J. R ; Sharif University of Technology
    IEEE  2000
    A wireless network with beamforming capabilities at the receiver is considered that allows two or more transmitters to share the same channel to communicate with the base station. A novel approach is introduced, which combines the effects of the digital signal processing (adaptive beamforming) at the physical layer with the traffic policies at the network layer on the overall queuing model of a cell. The effect of signal processing on the queuing model of the cell is represented by a parameter in the final cell model. Each cell is modeled by a multiuser/multiserver service facility, where each server is a beamformed channel formed by the cell's base station. From this effective cell model,... 

    Compressive detection of sparse signals in additive white Gaussian noise without signal reconstruction

    , Article Signal Processing ; Volume 131 , 2017 , Pages 376-385 ; 01651684 (ISSN) Hariri, A ; Babaie Zadeh, M ; Sharif University of Technology
    Elsevier B.V  2017
    The main motivation behind compressive sensing is to reduce the sampling rate at the input of a digital signal processing system. However, if for processing the sensed signal one requires to reconstruct the corresponding Nyquist samples, then the data rate will be again high in the processing stages of the overall system. Therefore, it is preferred that the desired processing task is done directly on the compressive measurements, without the need for the reconstruction of the Nyquist samples. This paper addresses the case in which the processing task is “detection” (the existence) of a sparse signal in additive white Gaussian noise, with applications e.g. in radar systems. Moreover, we will... 

    Receivers for diffusion-based molecular communication: Exploiting memory and sampling rate

    , Article IEEE Journal on Selected Areas in Communications ; Vol. 32, issue. 12 , 2014 , pp. 2368-2380 ; ISSN: 07338716 Mosayebi, R ; Arjmandi, H ; Gohari, A ; Nasiri-Kenari, M ; Mitra, U ; Sharif University of Technology
    In this paper, a diffusion-based molecular communication channel between two nano-machines is considered. The effect of the amount of memory on performance is characterized, and a simple memory-limited decoder is proposed; its performance is shown to be close to that of the best possible decoder (without any restrictions on the computational complexity or its functional form), using genie-aided upper bounds. This effect is adapted to the case of Molecular Concentration Shift Keying; it is shown that a four-bit memory achieves nearly the same performance as infinite memory for all of the examples considered. A general class of threshold decoders is considered and shown to be suboptimal for a... 

    Sparse signal processing using iterative method with adaptive thresholding (IMAT)

    , Article 2012 19th International Conference on Telecommunications, ICT 2012, 23 April 2012 through 25 April 2012, Jounieh ; 2012 ; 9781467307475 (ISBN) Marvasti, F ; Azghani, M ; Imani, P ; Pakrouh, P ; Heydari, S.J ; Golmohammadi, A ; Kazerouni, A ; Khalili, M. M ; Sharif University of Technology
    IEEE  2012
    Classical sampling theorem states that by using an anti-aliased low-pass filter at the Nyquist rate, one can transmit and retrieve the filtered signal. This approach, which has been used for decades in signal processing, is not good for high quality speech, image and video signals where the actual signals are not low-pass but rather sparse. The traditional sampling theorems do not work for sparse signals. Modern approach, developed by statisticians at Stanford (Donoho and Candes), give some lower bounds for the minimum sampling rate such that a sparse signal can be retrieved with high probability. However, their approach, using a sampling matrix called compressive matrix, has certain... 

    Enhanced TED: a new data structure for RTL verification

    , Article 21st International Conference on VLSI Design, VLSI DESIGN 2008, Hyderabad, 4 January 2008 through 8 January 2008 ; 2008 , Pages 481-486 ; 0769530834 (ISBN); 9780769530833 (ISBN) Lotfi Kamran, P ; Massoumi, M ; Mirzaei, M ; Navabi, Z ; VLSI Society of India ; Sharif University of Technology
    This work provides a canonical representation for manipulation of RTL designs. Work has already been done on a canonical and graph-based representation called Taylor Expansion Diagram (TED). Although TED can effectively be used to represent arithmetic expressions at the word-level, it is not memory efficient in representing bit-level logic expressions. In addition, TED cannot represent Boolean expressions at the word-level (vector-level). In this paper, we present modifications to TED that will improve its ability for bit-level logic representation while enhancing its robustness to represent word-level Boolean expressions. It will be shown that for bit-level logic expressions, the Enhanced... 

    Optimizing pipelines of trigonometric functions for FPGAs

    , Article 2007 IEEE Pacific Rim Conference on Communications, Computers and Signal Processing, PACRIM, Victoria, BC, 22 August 2007 through 24 August 2007 ; 2007 , Pages 105-108 ; 1424411904 (ISBN); 9781424411900 (ISBN) Ajorloo, H ; Ebrahimi, H ; Sarbazi Azad, H ; Sharif University of Technology
    Trigonometric functions are one of the most applicable functions in digital signal processing. In this paper, we propose two approaches for optimizing pipeline implementation of the CORDIC algorithm and compare it with other previous approaches. The proposed solutions are implemented on one of the Xilinx Virtex family's FPGAs. Our simulation results show that for high input bits, our approach is preferable to other existing approaches. ©2007 IEEE  

    Feasibility Study of TMS (DSP-Core Base)and Xilinx FPGA for Speech Algorithm

    , M.Sc. Thesis Sharif University of Technology Sabouri, Peyman (Author) ; Mortazavi, Mohammad (Supervisor) ; Ghorshi, Mohammad Ali (Supervisor)
    Digital Signal Processing (DSP) is used in a wide range of applications such as high-definition TV, digital audio, multimedia, digital cameras, radar, sonar detectors, biomedical imaging, global positioning, digital radio, speech recognition and etc. These applications can be implemented by either DSP processors or FPGA technology. Digital Signal Processors are microprocessors specifically designed to handle Digital Signal Processing tasks. These devices have seen a tremendous growth in the last decade, finding use in everything from cellular telephones to advanced scientific instruments. On the other hand, the rise of FPGA in the signal processing realm could be assigned to hardware to... 

    A parallel cepstral and spectral modeling for HMM-based speech enhancement

    , Article 17th DSP 2011 International Conference on Digital Signal Processing, Proceedings, 6 July 2011 through 8 July 2011, Corfu ; 2011 ; 9781457702747 (ISBN) Veisi, H ; Sameti, H ; Sharif University of Technology
    An HMM-based speech enhancement in Mel-frequency domain is introduced and improved. It is shown that hidden Markov modeling in the Mel-frequency domain is beneficial due to its effective representation of the speech spectrum; however, speech enhancement in this domain requires an inversion from the Mel-frequency to the spectral domain which introduces distortion artifacts for spectrum estimation. To reduce the distortion effects of the inversion and employ the advantages of robustness modeling in the Mel-frequency domain, a parallel cepstral and spectral (PCS) modeling is proposed. In PCS, a concurrent modeling in both cepstral and spectral domains is performed. The performances of the... 

    CMA-based adaptive antenna array digital beamforming with reduced complexity

    , Article 2010 7th International Symposium on Communication Systems, Networks and Digital Signal Processing, CSNDSP 2010, 21 July 2010 through 23 July 2010, Newcastle upon Tyne ; 2010 , Pages 327-331 ; 9781861353696 (ISBN) Shirvani Moghaddam, S ; Shirvani Moghaddam, M ; Kalami Rad, R ; Sharif University of Technology
    Reducing the computational complexity as well as convergence time is the main task of adaptive antenna array processing. This article proposes an improved algorithm for estimating the weights of adaptive array elements based on Constant Modulus Algorithm (CMA). In this new type of algorithm, those weights that have higher effect on the radiation pattern will be estimated and antenna pattern will be adjusted by changing these weights. In this research, 3 new algorithms are proposed. By simulating these algorithms and comparing them with conventional full weight CMA, finally new algorithm that has a reduced complexity and an acceptable performance at different signal to noise ratios (SNRs) is... 

    A novel adaptive LMS-based algorithm considering relative velocity of source

    , Article 2010 7th International Symposium on Communication Systems, Networks and Digital Signal Processing, CSNDSP 2010, 21 July 2010 through 23 July 2010, Newcastle upon Tyne ; 2010 , Pages 10-14 ; 9781861353696 (ISBN) Shirvani Moghaddam, S ; Shirvani Moghaddam, M ; Kalami Rad, R ; Sharif University of Technology
    In this paper a new least mean square (LMS) based adaptive weighting algorithm is proposed. It is appropriate for antenna array systems with moving targets and mobile applications. The essential goal of this algorithm is to reduce the complexity of weighting process and to decrease the time needed for adjusting the antenna radiation pattern. The main lobe of antenna will be adjusted in the direction of desired signal (main signal) and nulls pointed in the direction of undesired signals (interference signals). By predicting the relative velocity of source, the next location of the source will be estimated and the array weights will be determined using LMS algorithm before arriving to the new... 

    Approximateml estimator for compensation of timing mismatch and jitter noise in Ti-ADCS

    , Article European Signal Processing Conference, 28 August 2016 through 2 September 2016 ; Volume 2016-November , 2016 , Pages 2360-2364 ; 22195491 (ISSN) ; 9780992862657 (ISBN) Araghi, H ; Akhaee, M. A ; Amini, A ; Sharif University of Technology
    European Signal Processing Conference, EUSIPCO  2016
    Time-interleaved analog to digital converters (TI-ADC) offer high sampling rates by passing the input signal through C parallel low-rate ADCs. We can achieve C-times the sampling rate of a single ADC if all the shifts between the channels are identical. In practice, however, it is not possible to avoid mismatch among shifts. Besides, the samples are also subject to jitter noise. In this paper, we propose a blind method to mitigate the joint effects of sampling jitter and shift mismatch in the TI-ADC structure. We assume the input signal to be bandlimited and incorporate the jitter via a stochastic model. Next, we derive an approximate model based on a first-order Taylor series and use an... 

    Two quasi orthogonal space-time block codes with better performance and low complexity decoder

    , Article 10th International Symposium on Communication Systems, Networks and Digital Signal Processing, CSNDSP 2016, 20 July 2016 through 23 July 2016 ; 2016 ; 9781509025268 (ISBN) Lotfi Rezaabad, A ; Talebi, S ; Chizari, A ; Sharif University of Technology
    Institute of Electrical and Electronics Engineers Inc  2016
    This paper presents two new space time block codes (STBCs) with quasi orthogonal structure for an open loop multi-input single-output (MISO) systems. These two codes have been designed to transmit from three or four antennas at the transmitter and be given to one antenna at the receiver. In this paper first, the proposed codes are introduced and their structures are investigated. This is followed by the demonstration of how the decoder decodes half of transmitted symbols independent of the other half. The last part of this paper discusses the simulation results, makes performance comparison against other popular approaches and concludes that the proposed solutions offer superiority  

    Designing a dimmable OPPM-based VLC system under channel constraints

    , Article 10th International Symposium on Communication Systems, Networks and Digital Signal Processing, CSNDSP 2016, 20 July 2016 through 23 July 2016 ; 2016 ; 9781509025268 (ISBN) Chizari, A ; Jamali, M. V ; Abdollahramezani, S ; Salehi, J. A ; Dargahi, A ; Sharif University of Technology
    Institute of Electrical and Electronics Engineers Inc  2016
    In this paper, we design a dimming compatible visible light communication (VLC) system in a standard office room according to illumination standards under channel constraints. We use overlapping pulse position modulation (OPPM) to support dimming control by changing the code weights. The system parameters such as a valid interval for dimming together with an upper bound for bit rate according to the channel delay spread are investigated. Moreover, considering the dispersive VLC channel and using Monte Carlo (MC) simulations, a method is proposed to determine the minimum code length in different dimming levels in order to achieve a valid bit error rate (BER). Finally, trellis coded modulation...