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In high-speed analog-to-digital converters (ADCs), two main factors contribute to high power consumption. The first is the super linear relationship with the sampling rate; i.e., by doubling the sampling rate, the power consumption more than doubles. The second factor arises from the consumption of analog circuitry responsible to mitigate the jitter noise. By employing a multichannel sampling system, one can achieve high sampling rates by incorporating multiple low sampling-rate channels, which results in a linear scaling of power consumption with the number of channels. The main drawback of this system is the timing mismatch between the sampling channels. In this paper, we intend to jointly compensate the jitter noise and the timing mismatch between the channels using statistical methods. We first approximate the acquisition system and derive a stochastic model. Then, we propose an iterative maximum likelihood algorithm to estimate the parameters of the input signal. We further evaluate the Cramér-Rao lower bound (CRLB) for the estimation error to examine the proposed algorithm. Simulation results indicate that our algorithm is capable of closely following the CRLB curve for reasonable values of jitter noise and a wide range of timing mismatch errors. Moreover, it is shown that the mismatch-compensated multichannel sampling system performs almost equivalently to a single-channel high rate sampler without having its shortcomings. [ABSTRACT FROM AUTHOR]

This paper deals with area-based subpixel image registration under the rotation-isometric scaling-translation transformation hypothesis. Our approach is based on parametrical modeling of geometrically transformed textural image fragments and maximum-likelihood estimation of the transformation vector between them. Due to the parametrical approach based on the fractional Brownian motion modeling of the local fragments' texture, the proposed estimator \mboxML_\mathrmfBm (ML stands for “maximum likelihood” and fBm stands for “fractal Brownian motion”) has the ability to better adapt to real image texture content compared with other methods relying on universal similarity measures such as mutual information or normalized correlation. The main benefits are observed when assumptions underlying the fBm model are fully satisfied, e.g., for isotropic normally distributed textures with stationary increments. Experiments on both simulated and real images and for high and weak correlations between registered images show that the \mboxML_\mathrmfBm estimator offers significant improvement compared with other state-of-the-art methods. It reduces translation vector, rotation angle, and scaling factor estimation errors by a factor of about 1.75–2, and it decreases the probability of false match by up to five times. In addition, an accurate confidence interval for \mboxML_\mathrmfBm estimates can be obtained from the Cramér–Rao lower bound on rotation-scaling-translation parameter estimation error. This bound depends on texture roughness, noise level in reference and template images, correlation between these images, and geometrical transformation parameters. [ABSTRACT FROM AUTHOR]

Sensor-based online process monitoring has extensive applications, such as in manufacturing and service industries. In real environments, though, sensor data are often contaminated with noise, leading to severe challenges in accurate data analysis. In the existing literature, noise is generally modeled as Gaussian to analyze sensor data for various applications, for example in fault detection and diagnostics. However, in some applications, such as due to challenging field conditions, sensor data may be disturbed by high levels of outliers such that the Gaussian assumption of sensor noise is inadequate, thus leading to large estimation errors. This paper focuses on online classification applications. A robust sparse representation classification method is proposed, which considers non-Gaussian noise, and thus can effectively analyze sensor data with higher levels of outliers. Case studies were completed, based on both numerically simulated sensor data and actual wearable sensor data from occupational manual material handling process monitoring. The proposed classification method could effectively analyze sensor data with non-Gaussian noise, and outperformed commonly used methods in the literature. Thus, this new method may be advantageous for solving classification problems in challenging field conditions, to address the difficulties of high levels of sensor outliers. [ABSTRACT FROM AUTHOR]

Principal component analysis (PCA) aims at estimating the direction of maximal variability of a high-dimensional data set. A natural question is: does this task become easier, and estimation more accurate, when we exploit additional knowledge on the principal vector? We study the case in which the principal vector is known to lie in the positive orthant. Similar constraints arise in a number of applications, ranging from the analysis of gene expression data to spike sorting in neural signal processing. In the unconstrained case, the estimation performances of PCA have been precisely characterized using the random matrix theory, under a statistical model known as the spiked model. It is known that the estimation error undergoes a phase transition as the signal-to-noise ratio crosses a certain threshold. Unfortunately, tools from the random matrix theory have no bearing on the constrained problem. Despite this challenge, we develop an analogous characterization in the constrained case, within a one-spike model. In particular: 1) we prove that the estimation error undergoes a similar phase transition, albeit at a different thresholds in signal-to-noise ratio that we determine exactly; 2) we prove that—unlike in the unconstrained case—the estimation error depends on the spike vector, and characterize the least favorable vectors; and 3) we show that a non-negative principal component can be approximately computed—under the spiked model—in nearly linear time. This despite the fact that the problem is non-convex and, in general, NP-hard to solve exactly. [ABSTRACT FROM AUTHOR]

This letter presents a novel time-of-arrival (TOA) blind estimation technique for ultra wideband energy detection receivers with reduced complexity. The proposed method is blind in the sense that it does not exploit any information about channel or noise power. This new approach is based on a set of approximations of the exact likelihood function (ELF) of the observed energy. Even though these approximations achieve an important reduction of complexity, the shape of the new approximated function is accurate enough compared to the ELF. Application of a threshold to the differential of the approximated log-likelihood function completes the procedure. Simulations show that the performance of the proposed method in terms of the cumulative distribution function of the estimation error approaches that of a method based on the ELF and a genie-aided algorithm with perfect knowledge of the optimal threshold. [ABSTRACT FROM PUBLISHER]

We show that the rate regions achieved by linear-feedback coding schemes over dual multi-antenna Gaussian multi-access channels (MACs) and broadcast channels (BCs) with independent noises coincide. By dual here we mean: •

the channel matrices of the MAC and the BC are transposes of each other and [ABSTRACT FROM PUBLISHER]

This paper considers the problem of resource-constrained and noise-limited localization and estimation of dynamic targets that are sparsely distributed over a large area. We generalize an existing framework [Bashan , 2008] for adaptive allocation of sensing resources to the dynamic case, accounting for time-varying target behavior such as transitions to neighboring cells and varying amplitudes over a potentially long time horizon. The proposed adaptive sensing policy is driven by minimization of a surrogate function for mean squared error within locations containing targets. We provide theoretical upper bounds on the performance of adaptive sensing policies by analyzing solutions with oracle knowledge of target locations, gaining insight into the effect of target motion and amplitude variation as well as sparsity. Exact minimization of the multistage objective function is infeasible, but myopic optimization yields a closed-form solution. We propose a simple non-myopic extension, the Dynamic Adaptive Resource Allocation Policy (D-ARAP), that allocates a fraction of resources for exploring all locations rather than solely exploiting the current belief state. Our numerical studies indicate that D-ARAP has the following advantages: (a) it is more robust than the myopic policy to noise, missing data, and model mismatch; (b) it performs comparably to well-known approximate dynamic programming solutions but at significantly lower computational complexity; and (c) it improves greatly upon nonadaptive uniform resource allocation in terms of estimation error and probability of detection. [ABSTRACT FROM PUBLISHER]

This paper establishes a nearly optimal algorithm for denoising a mixture of sinusoids from noisy equispaced samples. We derive our algorithm by viewing line spectral estimation as a sparse recovery problem with a continuous, infinite dictionary. We show how to compute the estimator via semidefinite programming and provide guarantees on its mean-squared error rate. We derive a complementary minimax lower bound on this estimation rate, demonstrating that our approach nearly achieves the best possible estimation error. Furthermore, we establish bounds on how well our estimator localizes the frequencies in the signal, showing that the localization error tends to zero as the number of samples grows. We verify our theoretical results in an array of numerical experiments, demonstrating that the semidefinite programming approach outperforms three classical spectral estimation techniques. [ABSTRACT FROM PUBLISHER]

Minimum-variance filters and smoothers exhibit performance degradations when they are designed with inexact models and noise statistics. Filter and smoother estimation errors are assumed herein to be generated by a first-order moving-average system. This assumed system is identified and used to design a frequency weighting function to improve mean square error performance. It is shown under prescribed conditions that the sequence of frequency-weighted estimation error variances are nonincreasing. An example is presented which demonstrates the efficacy of repeated frequency weighting iterations. [ABSTRACT FROM AUTHOR]

This paper focuses on analytical studies of the primary user (PU) traffic classification problem. colorblackObserving that the gamma distribution can represent positively skewed data and exponential distribution (popular in communication networks performance analysis literature) it is considered here as the PU traffic descriptor. We investigate two PU traffic classifiers utilizing perfectly measured PU activity (busy) and inactivity (idle) periods: (i) maximum likelihood classifier (MLC) and (ii) multi-hypothesis sequential probability ratio test classifier (MSPRTC). Then, relaxing the assumption on perfect period measurement, we consider a PU traffic observation through channel sampling. For a special case of negligible probability of PU state change in between two samplings, we propose a minimum variance PU busy/idle period length estimator. Later, relaxing the assumption of the complete knowledge of the parameters of the PU period length distribution, we propose two PU traffic classification schemes: (i) estimate-then-classify (ETC), and (ii) average likelihood function (ALF) classifiers considering time domain fluctuation of the PU traffic parameters. Numerical results show that both MLC and MSPRTC are sensitive to the periods measurement errors when the distance among distribution hypotheses is small, and to the distribution parameter estimation errors when the distance among hypotheses is large. For PU traffic parameters with a partial prior knowledge of the distribution, the ETC outperforms ALF when the distance among hypotheses is small, while the opposite holds when the distance is large. [ABSTRACT FROM AUTHOR]

We study the theoretical performance of two full-duplex multiple-input multiple-output (MIMO) radio systems: a full-duplex bi-directional communication system and a full-duplex relay system. We focus on the effect of a (digitally manageable) residual self-interference due to imperfect channel estimation (with independent and identically distributed (i.i.d.) Gaussian channel estimation error) and transmitter noise. We assume that the instantaneous channel state information (CSI) is not available the transmitters. To maximize the system ergodic mutual information, which is a nonconvex function of power allocation vectors at the nodes, a gradient projection algorithm is developed to optimize the power allocation vectors. This algorithm exploits both spatial and temporal freedoms of the source covariance matrices of the MIMO links between transmitters and receivers to achieve higher sum ergodic mutual information. It is observed through simulations that the full-duplex mode is optimal when the nominal self-interference is low, and the half-duplex mode is optimal when the nominal self-interference is high. In addition to an exact closed-form ergodic mutual information expression, we introduce a much simpler asymptotic closed-form ergodic mutual information expression, which in turn simplifies the computation of the power allocation vectors. [ABSTRACT FROM PUBLISHER]

In this paper we present a generic framework for the asymptotic performance analysis of subspace-based parameter estimation schemes. It is based on earlier results on an explicit first-order expansion of the estimation error in the signal subspace obtained via an SVD of the noisy observation matrix. We extend these results in a number of aspects. Firstly, we demonstrate that an explicit first-order expansion of the Higher-Order SVD (HOSVD)-based subspace estimate can be derived. Secondly, we show how to obtain explicit first-order expansions of the estimation error of arbitrary ESPRIT-type algorithms and provide the expressions for R-D Standard ESPRIT, R-D Unitary ESPRIT, R-D Standard Tensor-ESPRIT, as well as R-D Unitary Tensor-ESPRIT. Thirdly, we derive closed-form expressions for the mean square error (MSE) and show that they only depend on the second-order moments of the noise. Hence, to apply this framework we only need the noise to be zero mean and possess finite second order moments. Additional assumptions such as Gaussianity or circular symmetry are not needed. [ABSTRACT FROM PUBLISHER]

In this correspondence, we propose an efficient estimator of optimal memory (averaging interval) for discrete-time finite impulse response (FIR) filters in state-space. Its crucial property is that only real measurements and the filter output are involved with no reference and noise statistics. Testing by the two-state polynomial model has shown a very good correspondence with predicted values. Even in the worst case of the harmonic model, the estimator demonstrates practical applicability. [ABSTRACT FROM AUTHOR]

The accurate determination of the steering vector of a sensor array that corresponds to a desired signal is often hindered by uncertainties due to array imperfections, such as the presence of a direction-of-arrival (DOA) estimation error, mutual coupling, array sensor gain/phase uncertainties, and senor position perturbations. Consequently, the performance of conventional beamforming algorithms that use the nominal steering vector may be significantly degraded. A new method for recursively correcting possible deterministic errors in the estimated steering vector is proposed here. It employs the subspace principle and estimates the desired steering vector by using a convex optimization approach. We show that the solution can be obtained in closed form by using the Lagrange multiplier method. As the proposed method is based on an extended version of the conventional orthonormal PAST (OPAST) algorithm, it has low implementation complexity, and moving sources can be handled. In addition, a robust beamformer with a new error bound that uses the proposed steering vector estimate is derived by optimizing the worst case performance of the array after taking the uncertainties of the array covariance matrix into account. This gives a diagonally loaded Capon beamformer, where the loading level is related to the bound of the uncertainty in the array covariance matrix. Numerical results show that the proposed algorithm performs well, especially at high signal-to-noise ratio (SNR) and in the presence of deterministic sensor uncertainties. [ABSTRACT FROM PUBLISHER]

This paper proposes a Kalman filter-based attitude (i.e., roll and pitch) estimation algorithm using an inertial sensor composed of a triaxial accelerometer and a triaxial gyroscope. In particular, the proposed algorithm has been developed for accurate attitude estimation during dynamic conditions, in which external acceleration is present. Although external acceleration is the main source of the attitude estimation error and despite the need for its accurate estimation in many applications, this problem that can be critical for the attitude estimation has not been addressed explicitly in the literature. Accordingly, this paper addresses the combined estimation problem of the attitude and external acceleration. Experimental tests were conducted to verify the performance of the proposed algorithm in various dynamic condition settings and to provide further insight into the variations in the estimation accuracy. Furthermore, two different approaches for dealing with the estimation problem during dynamic conditions were compared, i.e., threshold-based switching approach versus acceleration model-based approach. Based on an external acceleration model, the proposed algorithm was capable of estimating accurate attitudes and external accelerations for short accelerated periods, showing its high effectiveness during short-term fast dynamic conditions. Contrariwise, when the testing condition involved prolonged high external accelerations, the proposed algorithm exhibited gradually increasing errors. However, as soon as the condition returned to static or quasi-static conditions, the algorithm was able to stabilize the estimation error, regaining its high estimation accuracy. [ABSTRACT FROM AUTHOR]

Numerous variable step-size normalized least mean-square (VSS-NLMS) algorithms have been derived to solve the dilemma of fast convergence rate or low excess mean-square error in the past two decades. This paper proposes a new, easy to implement, nonparametric VSS-NLMS algorithm that employs the mean-square error and the estimated system noise power to control the step-size update. Theoretical analysis of its steady-state behavior shows that, when the input is zero-mean Gaussian distributed, the misadjustment depends only on a parameter \beta controlling the update of step size. Simulation experiments show that the proposed algorithm performs very well. Furthermore, the theoretical steady-state behavior is in very good agreement with the experimental results. [ABSTRACT FROM PUBLISHER]

This paper considers linear minimum mean-square-error (MMSE) transceiver design problems for downlink multiuser multiple-input multiple-output (MIMO) systems where imperfect channel state information is available at the base station (BS) and mobile stations (MSs). We examine robust sum mean-square-error (MSE) minimization problems. The problems are examined for the generalized scenario where the power constraint is per BS, per BS antenna, per user or per symbol, and the noise vector of each MS is a zero-mean circularly symmetric complex Gaussian random variable with arbitrary covariance matrix. For each of these problems, we propose a novel duality based iterative solution. Each of these problems is solved as follows. First, we establish a novel sum average mean-square-error (AMSE) duality. Second, we formulate the power allocation part of the problem in the downlink channel as a Geometric Program (GP). Third, using the duality result and the solution of GP, we utilize alternating optimization technique to solve the original downlink problem. To solve robust sum MSE minimization constrained with per BS antenna and per BS power problems, we have established novel downlink-uplink duality. On the other hand, to solve robust sum MSE minimization constrained with per user and per symbol power problems, we have established novel downlink-interference duality. For the total BS power constrained robust sum MSE minimization problem, the current duality is established by modifying the constraint function of the dual uplink channel problem. And, for the robust sum MSE minimization with per BS antenna and per user (symbol) power constraint problems, our duality are established by formulating the noise covariance matrices of the uplink and interference channels as fixed point functions, respectively. We also show that our sum AMSE duality are able to solve other sum MSE-based robust design problems. Computer simulations verify the robustness of the proposed robust designs compared to the nonrobust/naive designs. [ABSTRACT FROM PUBLISHER]

This paper considers the problem of adaptive array processing for interference canceling to drive very deep nulls in difficult signal environments. In many practical scenarios, the achievable null depth is limited by covariance matrix estimation error leading to poor identification of the interference subspace. We address the particularly troublesome cases of low interference-to-noise ratio (INR), relatively rapid interference motion, and correlated noise across the receiving array. A polynomial-based model is incorporated in the proposed algorithm to track changes in the array covariance matrix over time, mitigate interference subspace estimation errors, and improve canceler performance. The application of phased array feeds for radio astronomical telescopes is used to illustrate the problem and proposed solution. Here even weak residual interference after cancelation may obscure a signal of interest, so very deep beampattern nulls are required. Performance for conventional subspace projection (SP) is compared with polynomial-augmented SP using simulated and real experimental data, showing null-depth improvement of 6 to 30 dB. [ABSTRACT FROM PUBLISHER]

The performance of narrowband active noise control (ANC) systems degenerates significantly under the frequency mismatch between the synthesized reference signal and the primary noise. This paper proposes a new narrowband ANC system that incorporates a frequency corrector for the internal reference signal generator to mitigate the ill-effects caused by the frequency mismatch. The proposed frequency estimator uses the iterative frequency estimation algorithm based on the minimum variance distortionless response spectrum that has advantages of accuracy and fast convergence. The performance of the new narrowband ANC system is analyzed under the conditions of frequency mismatch and the secondary-path phase estimation error. The frequency and amplitude of the primary noise are modeled as time-varying variables in the theoretical analysis. Computer simulations are conducted to demonstrate the performance of the new ANC system and to verify the analysis results. [ABSTRACT FROM PUBLISHER]

Quantization of phase and/or amplitude has far-reaching effects on the radiation characteristics of the phased array (PA), including gain, minor lobe level, and point deviation. Traditionally, one common method to address such a nonlinear distortion is using random-phasing to interrupt the error periodicity. Here, we show that the distortion due to the quantization can be better remediated by spectrally shaping the error compared to the random-phasing (dithering) approaches. We adapted the method for phase-only and amplitude-phase synthesis of planar array designed based on analog beamforming (ABF). To do that, for the first time, 2-D real- and complex-coefficient minimum-phase digital finite impulse response (FIR) filters are designed based on the discrete Hilbert transform (DHT) method. In particular, the digital filter design for phase-only synthesis is comprehensively investigated, respecting the error spectra in the beamspace domain. It is shown that by pushing the error out of the so-called visible region, the decrease of antenna directivity due to the quantization can be compensated to some extent, which provides a quite advantage over the uniform distribution of error. For some cases, pushing the error out of the visible region might be impossible. For such cases, we proposed using the spaced-notches filter. It is also shown that the method is on maximum efficacy when both phase and amplitude of the excitation signal are controllable. Thus, complex-valued noise shaping (CV-NS) can be exploited for the phase-amplitude synthesis of the PA, showing quite promising performance. [ABSTRACT FROM AUTHOR]

This article analyzes the propagation of feedback noise in multisampled dc–dc power converters. The analytical calculations for noise attenuation, strictly valid for linear time-invariant systems, are found to offer good predictions for power converters, after suppressing the decimation effects of the digital pulsewidth modulator (DPWM). For control systems that employ a proportional-integral controller, without any digital low-pass filters, the nonlinearity caused by the DPWM decimation is found to saturate the noise attenuation properties, as the multisampling factor $N$ is increased. Strong noise attenuation is enabled using antialiasing digital filters, with a small impact on the dynamic response. For control systems that employ a proportional-integral-derivative controller, increase of the multisampling factor, without any digital filters, causes the amplification of the noise power. Therefore, a greater attenuation of high-frequency components is required to provide a significant noise reduction in the control bandwidth. Even with these antialiasing digital filters, the dynamic response of multisampled control systems is improved compared to double-update. This is proven by the analytical and experimental comparison of various multisampled control strategies in terms of dynamic response and noise attenuation capabilities. The experimental results, from a buck-type converter, match well with simulations and analytical calculations. [ABSTRACT FROM AUTHOR]

In this letter, we propose a robust algorithm for determining a unitary transform of a discrete-time signal, whose samples are sensed in impulsive noise environments. The proposed algorithm estimates the transformed signal coefficients by solving an ℓ1-regularized least absolute deviation (ℓ1-LAD) regression problem, leading to the weighted median (WM) as the optimal operator for computing each transform coefficient. Numerical simulations on synthetic data are presented in order to compare the performance of the proposed algorithm to those yielded by previously reported methods. Furthermore, an example of audio denoising using the short time Fourier transform of a digital audio record, in the presence of impulsive noise, is also shown. [ABSTRACT FROM AUTHOR]

Traditionally, the hinge loss is used to construct support vector machine (SVM) classifiers. The hinge loss is related to the shortest distance between sets and the corresponding classifier is hence sensitive to noise and unstable for re-sampling. In contrast, the pinball loss is related to the quantile distance and the result is less sensitive. The pinball loss has been deeply studied and widely applied in regression but it has not been used for classification. In this paper, we propose a SVM classifier with the pinball loss, called pin-SVM, and investigate its properties, including noise insensitivity, robustness, and misclassification error. Besides, insensitive zone is applied to the pin-SVM for a sparse model. Compared to the SVM with the hinge loss, the proposed pin-SVM has the same computational complexity and enjoys noise insensitivity and re-sampling stability. [ABSTRACT FROM PUBLISHER]

This paper presents a novel framework to uncertain one-class learning and concept summarization learning on uncertain data streams. Our proposed framework consists of two parts. First, we put forward uncertain one-class learning to cope with data of uncertainty. We first propose a local kernel-density-based method to generate a bound score for each instance, which refines the location of the corresponding instance, and then construct an uncertain one-class classifier (UOCC) by incorporating the generated bound score into a one-class SVM-based learning phase. Second, we propose a support vectors (SVs)-based clustering technique to summarize the concept of the user from the history chunks by representing the chunk data using support vectors of the uncertain one-class classifier developed on each chunk, and then extend k-mean clustering method to cluster history chunks into clusters so that we can summarize concept from the history chunks. Our proposed framework explicitly addresses the problem of one-class learning and concept summarization learning on uncertain one-class data streams. Extensive experiments on uncertain data streams demonstrate that our proposed uncertain one-class learning method performs better than others, and our concept summarization method can summarize the evolving interests of the user from the history chunks. [ABSTRACT FROM AUTHOR]

In this letter, we address the problem of blindly reconstructing compressively sensed signals by exploiting the co-sparse analysis model. In the analysis model it is assumed that a signal multiplied by an analysis operator results in a sparse vector. We propose an algorithm that learns the operator adaptively during the reconstruction process. The arising optimization problem is tackled via a geometric conjugate gradient approach. Different types of sampling noise are handled by simply exchanging the data fidelity term. Numerical experiments are performed for measurements corrupted with Gaussian as well as impulsive noise to show the effectiveness of our method. [ABSTRACT FROM PUBLISHER]

This paper describes a task-based, information-theoretic approach to the assessment of image quality in diagnostic sonography. We expand the Kullback–Leibler divergence metric J, which quantifies the diagnostic information contained within recorded radio-frequency echo signals, into a spatial-frequency integral comprised of two spectral components: one describes patient features for low-contrast diagnostic tasks and the other describes instrumentation properties. The latter quantity is the acquisition information spectrum (AIS), which measures the density of object information that an imaging system is able to transfer to the echo data at each spatial frequency. AIS is derived based on unique properties of acoustic scattering in tissues that generate object contrast. Predictions made by the J integral expression were validated through Monte Carlo studies using echo-signal data from simulated lesions. Our analysis predicts the diagnostic performance of any sonographic system at specific diagnostic tasks based on engineering properties of the instrument that constitute image quality. [ABSTRACT FROM AUTHOR]

This paper presents a frequency-translating bandpass \Delta \Sigma modulator that uses single-path mixing within the \Delta \Sigma feedback loop to downconvert a 4-MHz signal band from IF1 = 225 MHz at the input to IF2 = \ 25 MHz at the output. The proposed \Delta \Sigma modulator is designed with a sixth-order continuous-time loop filter and a 3-bit quantizer. The quantizer operates at a sampling frequency of 100 MHz, which is lower than IF1 and, therefore, reduces both the power consumption and the sensitivity to timing errors relative to a conventional bandpass \Delta \Sigma modulator. Furthermore, the loop filter implements noise shaping primarily at IF 2, which reduces the sensitivity to coefficient variations. The prototype chip was fabricated in 65-nm CMOS with an active area of 0.55 mm^2. It achieves an SNDR of 55 dB over a 4-MHz signal bandwidth, and consumes 13 mW from a 1-V power supply. [ABSTRACT FROM AUTHOR]

Timing compensation between the clock period and datapath delay in the presence of resonant supply noise has drawn a great deal of attention from the circuit design community. This effect, which is often referred to as the clock data compensation effect, manifests itself as an increase in maximum operating frequency for high performance microprocessors. In this work, we propose an adaptive phase-shifting PLL that can achieve optimal clock data compensation by digitally programming the supply noise sensitivity and the phase shift of the PLL clock period. Measurement results from a 1.2 V, 65 nm test chip demonstrate a 3.4–7.3% improvement in the maximum operating frequency across different clock distribution designs and resonant frequencies. A mathematical framework for simulating the performance of the adaptive phase-shifting PLL is presented for better insight on how the proposed PLL performs when used in different clock network configurations. In addition, the impact of the proposed technique on PLL stability as well as its effectiveness in a 32 nm process has been explored. [ABSTRACT FROM AUTHOR]

We analyze the operation and performance of subsampled analog optical links. These links downconvert microwave signals by optically sampling at a rate much lower than the RF carrier frequency. Expressions for conversion gain, noise figure, and spurious-free dynamic range are derived for intensity modulation and direct detection. The impact of pulse width on link performance is revealed and quantified. Also, in order to characterize noise in a subsampled link, the influence of timing jitter in the periodic optical source is considered. Timing jitter and pulse width are found to be two important factors which limit the noise figure and dynamic range in subsampled links. [ABSTRACT FROM PUBLISHER]

Assuming that the network delays are normally distributed and the network nodes are subject to clock phase offset errors, the maximum likelihood estimator (MLE) and the Kalman filter (KF) have been recently proposed with the goal of maximizing the clock synchronization accuracy in wireless sensor networks (WSNs). However, because the network delays may assume any distribution and the performance of MLE and KF is quite sensitive to the distribution of network delays, designing clock synchronization algorithms that are robust to arbitrary network delay distributions appears as an important problem. Adopting a Bayesian framework, this paper proposes a novel clock synchronization algorithm, called the Iterative Gaussian mixture Kalman particle filter (IGMKPF), which combines the Gaussian mixture Kalman particle filter (GMKPF) with an iterative noise density estimation procedure to achieve robust performance in the presence of unknown network delay distributions. The Kullback-Leibler divergence is used as a measure to assess the departure of estimated observation noise density from its true expression. The posterior Cramér–Rao bound (PCRB) and the mean-square error (MSE) of IGMKPF are evaluated via computer simulations. It is shown that IGMKPF exhibits improved performance and robustness relative to MLE. The prior information plays an important role in IGMKPF. A MLE-based method for obtaining reliable prior information for clock phase offsets is presented and shown to ensure the convergence of IGMKPF. [ABSTRACT FROM AUTHOR]

Successive approximation register (SAR) ADC architectures are popular for achieving high energy efficiency but they suffer from resolution and speed limitations. On the other hand pipeline ADC architectures can achieve high resolution and speed but have lower energy-efficiency and are more complex. We propose a two-stage pipeline ADC architecture with a large first-stage resolution, enabled with the help of a SAR-based sub-ADC. The prototype 12b 50 MS/s ADC achieves an ENOB of 10.4b at Nyquist, and a figure-of-merit of 52 fJ/conversion-step. The ADC achieves low-power, high-resolution and high-speed operation without calibration. The ADC is fabricated in 65 nm and 90 nm CMOS and occupies a core area of only 0.16\ \ mm^2. [ABSTRACT FROM AUTHOR]

In order to reduce the inter-carrier interference (ICI) produced by frequency offset in OFDM systems, we set up an optimization problem to find the transmission pulse which maximizes the signal-to-average-ICI power ratio. Furthermore, by solving a constrained convex optimization problem, our pulses can satisfy various decay requirements, for instance with respect to a prescribed spectral mask. The latter is an important feature in order to comply with industry standard specifications. Furthermore we discuss issues concerning the numerical treatment of the constrained convex optimization problem. Simulation results show that our pulses outperform many currently known transmission pulses with respect to the ICI criterion. [ABSTRACT FROM AUTHOR]

Many noise sources deteriorate the performance of a Fourier-transform spectrometer. When such an instrument utilizes a reference source such as an HeNe laser to sample the measurement data, it is desirable to determine how the intensity fluctuations of the reference source will affect the final spectrum. This text shows how the intensity noise of the reference laser is transformed at the output of the interferometer, how it translates into a sampling jitter by a simple linear relation under a small bandwidth approximation, and how it is manifested in the final spectrum as a noise floor. Analytical and simulation results show that for an instrument designed for optical communication applications, the intensity noise of the reference laser may be one of the dominant noise source limiting the overall performance of the instrument. [ABSTRACT FROM AUTHOR]

In this paper, the performance of a code-division multiple-access (CDMA) system using digital frequency conversion is analyzed and modeled. The joint effect of channel noise, sampling jitter, and analog-digital converter (ADC) errors is discussed and modeled assuming uniform converters. Variable resolution memoryless and ΣΔ ADCs are considered, and the sensitivities of the resulting systems to channel noise, sampling jitter, and overloading errors are compared. [ABSTRACT FROM AUTHOR]

On-chip interconnection networks for future systems on chip (SoC) will have to deal with the increasing sensitivity of global wires to noise sources such as crosstalk or power supply noise. Hence, transient delay and logic faults are likely to reduce the reliability of across-chip communication. Given the reduced power budgets for SoCs, in this paper, we develop solutions for combined energy minimization and communication reliability control. Redundant bus coding is proved to be an effective technique for trading off energy against reliability, so that the most efficient scheme can be selected to meet predefined reliability requirements in a low signal-to-noise ratio regime. We model on-chip interconnects as noisy channels and evaluate the impact of two error recovery schemes on energy efficiency: correction at the receiver stage versus retransmission of corrupted data. The analysis is performed in a realistic SoC setting, and holds both for shared communication resources and for peer-to-peer links in a network of interconnects. We provide SoC designers with guidelines for the selection of energy efficient error-control schemes for communication architectures. [ABSTRACT FROM AUTHOR]

Presents methods that provide channel estimates for a successful operation of blanking compensation algorithm. Details of the limitation and performances of various approaches in terms of mean squared error and bit error rate ratio; Information on impulse noise and channel models; Results on the estimation of burst position.

Presents an approach to adaptively match filter the I and Q components of complex-valued inputs consisting of a desired signal embedded in correlated external noise. Technique for I, Q adaptive match filtering (AMF); Example to demonstrate the utility of the almost phase invariant I, Q AMF; Conclusions.