Sharif Digital Repository

In this paper, an attempt to identify and characterize minor road traffic behavior at unsignalized urban intersections in a city situated in a developing country is described. Information about the particular traffic behavior was collected using video cameras placed at 12 unsignalized intersections connecting minor to major roadways in Tehran. Stop signs and/or blinking red signals provided the stopping message at the minor approaches of the selected intersections. Relevant information about the behavior of more than 2400 vehicles was extracted from the video displays and the database records consisted of 31 variables reflecting driver, pedestrian, passenger, vehicle and intersection... 

In this paper, the results of a study comparing the performance of two types of interchange, i.e., Conventional Diamond Interchange and Single Point Urban Diamond Interchange, are reported. The analysis is based on simulation of a Single Point Urban Diamond Interchange. The real data that is collected from two different existing Conventional Diamond Interchanges during peak hours is used as input for analysis. Highway capacity software is used to analyze the performance of the Conventional Diamond Interchanges while a simulation model is used for analyzing the performance of the Single Point Urban Diamond Interchanges. The results indicate that the performance of the Single Point Urban... 

Channel coding has been considered as a classical approach to overcome corruptions occurring in some elements of input signal which may lead to loss of some information. Proper redundancies are added to the input signal to improve the capability of detecting or even correcting the corrupted signal. A similar scenario may happen dealing with real-field numbers rather than finite-fields. This paper considers a way to reconstruct an exact version of a corrupted signal by using an encoded signal with proper number of redundancies. The proposed algorithm uses Graduated Non-Convexity method beside using a smoothed function instead of 0-norm to correct all the corrupted elements. Simulations show... 

This paper presents a new approach for hiding information in speech signals. In this method, the silence intervals of speech are found and the length (number of samples) of these intervals is changed to hide information. This method can be used simultaneously with other methods. © 2008 IEEE  

In this paper several applications of the Independent Component Analysis (ICA) algorithm, for the analysis of biomedical signal recordings have been investigated. One of these applications is the removal of EEG artifacts such as the EOG. It is shown that ICA may serve as a powerful tool, which could help the analysis of biomedical recordings, and give better insights about the underlying sources of some disorders. Another application of the proposed method is the detection of sleep disorders in patients suffering from sleep apnea. The ultimate goal of this approach is to develop an automatic noninvasive data acquisition system, for clinical applications  

In this paper we introduce a dynamical model for Phonocardiogram (PCG) signal which is capable of generating realistic synthetic PCG signals. This model is based on PCG morphology and consists of three ordinary differential equations and can represent various morphologies of normal PCG signals. Beat-to-beat variation in PCG morphology is significant so model parameters vary from beat to beat. This model is inspired of Electrocardiogram (ECG) dynamical model proposed by McSharry et al. and can be employed to assess biomedical signal processing techniques  

In Analog to Digital Converters, the sampled values are not necessarily sampled in the desired time. Sampling may occur before or after the desired time because of using non ideal converters; therefore the output signal of converters contains a noise related to this inexact sampling time. In this paper, signal to noise ratio according to practical sampling of continuous signal will be estimated. For this estimation, we have assumed the continuous signal as a band limited, zero-mean real signal which is wide sense stationary (wss). Sampling rate is also considered equal or more than Nyquist sampling rate. It will be concluded that what condition should be applied in order to have any... 

A modular method was suggested before to recover a band limited signal from the sample and hold and linearly interpolated (or, in general, an nth-order-hold) version of the regular samples. In this paper a novel approach for compensating the distortion of any interpolation based on modular method has been proposed. In this method the performance of the modular method is optimized by adding only some simply calculated coefficients. This approach causes drastic improvement in terms of signal-to-noise ratios with fewer modules compared to the classical modular method. Simulation results clearly confirm the improvement of the proposed method and also its superior robustness against additive... 

In some applications, the variance of measurement noise depends on the signal that we aim to measure. For instance, additive Gaussian signal-dependent noise (AGSDN) channel models are used in molecular and optical communication. Herein we provide lower and upper bounds on the capacity of additive signal dependent noise (ASDN) channels. We also provide sufficient conditions under which the capacity becomes infinity. © 2017 IEEE  

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  

Discrete multitone (DMT) modulation is used for the efficient transmission of data over nonideal channels. By nonideal channel, we mean that the channel with the frequency response is not flat. Also the noise is present in this kind of channel. In recent years, optimization of filter banks for the DMT systems is considered. Recent researches focused on the orthonormal filter banks have shown that the principal component filter bank (PCFB) is optimal for this application. We focus on one class of biorthogonal filter banks and find the optimum solution for this class. We also have shown that the performance of this structure is better than the optimum orthonormal filter bank. © 2003 IEEE  

The authors formulate the transportation mixed network design problem (MNDP) as a mixed-integer bi-level mathematical problem, based on the concept of reserve capacity. The upper level goal is to maximize the reserve capacity by signal settings at intersections, determine street direction and increase street capacities via addition of lanes. The lower level problem is a deterministic user equilibrium traffic assignment problem to minimize the user travel time. The model being non-convex, meta-heuristic methods are used to solve the problem. A hybridization of genetic algorithm with simulated annealing and a bee algorithm are proposed. Numerical examples are illustrated to verify the... 

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  

Echo cancellers with long impulse responses are usually used in networks with long echoes. Since long adaptive filters with sparse target filters usually converge very slowly, adaptation algorithms with fast convergence rate are needed. PNLMS [2] is one of the algorithms that is designed for fast convergence on sparse impulse responses. But there are some disadvantages to this algorithm. The computational complexity of this algorithm is prohibitive, especially for long echo tails. Another disadvantage is that its convergence rate slows down significantly after the adaptation of large taps. In this paper we propose new algorithms to solve both these problems. In another part of this paper, we... 

Autocorrelation receiver (AcR) is usually used as the general receiver for transmitted reference (TR) communication systems on a quite intuitive basis. In this paper, we apply the principle of statistical invariance to obtain the optimum TR receiver on a decision-theoretic basis, and it is shown that the autocorrelation receiver is optimal only in a special case. Performance of the optimum receiver and the suboptimum AcR for some important modulation schemes is also evaluated. The results imply that allocating a part of the transmitted energy and bandwidth to reference pulses would not help improve the performance of the system and the optimal performance might be achieved via much simpler... 

In this paper, we introduce a model-based Bayesian denoising framework for phonocardiogram (PCG) signals. The denoising framework is founded on a new dynamical model for PCG, which is capable of generating realistic synthetic PCG signals. The introduced dynamical model is based on PCG morphology and is inspired by electrocardiogram (ECG) dynamical model proposed by McSharry et al. and can represent various morphologies of normal PCG signals. The extended Kalman smoother (EKS) is the Bayesian filter that is used in this study. In order to facilitate the adaptation of the denoising framework to each input PCG signal, the parameters are selected automatically from the input signal itself. This... 

In this paper an iterative method for recovery of the missing samples of signals is investigated in detail, and some novel linear, nonlinear, and adaptive extrapolation techniques are proposed to be used along with it to increase the convergence rate of the recovery system. The proposed methods would remarkably speed up the convergence rate, save processing power, and reduce the delay of the system compared to some well known accelerated versions of the aforementioned iterative algorithm. © 2007 Elsevier B.V. All rights reserved  

In this paper, an approximate analysis for SNR improvement factor in non-coherent integrators for communication channels is introduced. The SNR improvement factor and loss function of the non-coherent integrator are calculated. Our results are very simple and illustrative. Although calculations are with some approximations and therefore the results are not exact, but the error is negligible, as it is shown. © 2005 IEEE  

In this paper an Extended Kalman Filter (EKF) has been proposed for the filtering of noisy ECG signals. The method is based on a modified nonlinear dynamic model, previously introduced for the generation of synthetic ECG signals. An automatic parameter selection method has also been suggested, to adapt the model with a vast variety of normal and abnormal ECG signals. The results show that the EKF output is able to track the original ECG signal shape even in the most noisiest epochs of the ECG signal. The proposed method may serve as an efficient filtering procedure for applications such as the noninvasive extraction of fetal cardiac signals from maternal abdominal signals. © 2005 IEEE  

Spectral subtraction is one of the most famous and common-used methods for speech enhancement. The main weakness of this method is the production of an annoying noise called musical noise. In this paper, we have reduced the musical noise and improved the quality of enhanced speech by increasing the accuracy of the system spectral estimator. This method is useful for speech enhancement systems in which the speech signal is degraded by stationary or near-stationary noises. Different experimental results in different SNRs confirmed the better performance of our system from both objective and subjective views, which means better quality and more SNR improvement. © 2005 IEEE