Biblio
Compressed sensing (CS) integrates sampling and compression into a single step to reduce the processed data amount. However, the CS reconstruction generally suffers from high complexity. To solve this problem, compressive signal processing (CSP) is recently proposed to implement some signal processing tasks directly in the compressive domain without reconstruction. Among various CSP techniques, compressive detection achieves the signal detection based on the CS measurements. This paper investigates the compressive detection problem of random signals when the measurements are corrupted. Different from the current studies that only consider the dense noise, our study considers both the dense noise and sparse error. The theoretical performance is derived, and simulations are provided to verify the derived theoretical results.
Compressed sensing can represent the sparse signal with a small number of measurements compared to Nyquist-rate samples. Considering the high-complexity of reconstruction algorithms in CS, recently compressive detection is proposed, which performs detection directly in compressive domain without reconstruction. Different from existing work that generally considers the measurements corrupted by dense noises, this paper studies the compressive detection problem when the measurements are corrupted by both dense noises and sparse errors. The sparse errors exist in many practical systems, such as the ones affected by impulse noise or narrowband interference. We derive the theoretical performance of compressive detection when the sparse error is either deterministic or random. The theoretical results are further verified by simulations.
Compressed Sensing or Compressive Sampling is the process of signal reconstruction from the samples obtained at a rate far below the Nyquist rate. In this work, Differential Pulse Coded Modulation (DPCM) is coupled with Block Based Compressed Sensing (CS) reconstruction with Robbins Monro (RM) approach. RM is a parametric iterative CS reconstruction technique. In this work extensive simulation is done to report that RM gives better performance than the existing DPCM Block Based Smoothed Projected Landweber (SPL) reconstruction technique. The noise seen in Block SPL algorithm is not much evident in this non-parametric approach. To achieve further compression of data, Lempel-Ziv-Welch channel coding technique is proposed.
Compressive Sampling and Sparse reconstruction theory is applied to a linearly frequency modulated continuous wave hybrid lidar/radar system. The goal is to show that high resolution time of flight measurements to underwater targets can be obtained utilizing far fewer samples than dictated by Nyquist sampling theorems. Traditional mixing/down-conversion and matched filter signal processing methods are reviewed and compared to the Compressive Sampling and Sparse Reconstruction methods. Simulated evidence is provided to show the possible sampling rate reductions, and experiments are used to observe the effects that turbid underwater environments have on recovery. Results show that by using compressive sensing theory and sparse reconstruction, it is possible to achieve significant sample rate reduction while maintaining centimeter range resolution.
This paper reviews some existing Speech Enhancement techniques and also proposes a new method for enhancing the speech by combining Compressed Sensing and Kalman filter approaches. This approach is based on reconstruction of noisy speech signal using Compressive Sampling Matching Pursuit (CoSaMP) algorithm and further enhanced by Kalman filter. The performance of the proposed method is evaluated and compared with that of the existing techniques in terms of intelligibility and quality measure parameters of speech. The proposed algorithm shows an improved performance compared to Spectral Subtraction, MMSE, Wiener filter, Signal Subspace, Kalman filter in terms of WSS, LLR, SegSNR, SNRloss, PESQ and overall quality.
This paper presents a model calibration algorithm for the modulated wideband converter (MWC) with non-ideal analog lowpass filter (LPF). The presented technique uses a test signal to estimate the finite impulse response (FIR) of the practical non-ideal LPF, and then a digital compensation filter is designed to calibrate the approximated FIR filter in the digital domain. At the cost of a moderate oversampling rate, the calibrated filter performs as an ideal LPF. The calibrated model uses the MWC system with non-ideal LPF to capture the samples of underlying signal, and then the samples are filtered by the digital compensation filter. Experimental results indicate that, without making any changes to the architecture of MWC, the proposed algorithm can obtain the samples as that of standard MWC with ideal LPF, and the signal can be reconstructed with overwhelming probability.
In view of the difficulty in selecting wavelet base and decomposition level for wavelet-based de-noising method, this paper proposes an adaptive de-noising method based on Ensemble Empirical Mode Decomposition (EEMD). The autocorrelation, cross-correlation method is used to adaptively find the signal-to-noise boundary layer of the EEMD in this method. Then the noise dominant layer is filtered directly and the signal dominant layer is threshold de-noised. Finally, the de-noising signal is reconstructed by each layer component which is de-noised. This method solves the problem of mode mixing in Empirical Mode Decomposition (EMD) by using EEMD and combines the advantage of wavelet threshold. In this paper, we focus on the analysis and verification of the correctness of the adaptive determination of the noise dominant layer. The simulation experiment results prove that this de-noising method is efficient and has good adaptability.
Sampling and reconstruction (S&R) are used in virtually all areas of science and technology. The classical sampling theorem is a theoretical foundation of S&R. However, for a long time, only sampling rates and ways of the sampled signals representation were derived from it. The fact that the design of S&R circuits (SCs and RCs) is based on a certain interpretation of the sampling theorem was mostly forgotten. The traditional interpretation of this theorem was selected at the time of the theorem introduction because it offered the only feasible way of S&R realization then. At that time, its drawbacks did not manifest themselves. By now, this interpretation has largely exhausted its potential and inhibits future progress in the field. This tutorial expands the theoretical foundation of S&R. It shows that the traditional interpretation, which is indirect, can be replaced by the direct one or by various combinations of the direct and indirect interpretations that enable development of novel SCs and RCs (NSCs and NRCs) with advanced properties. The tutorial explains the basic principles of the NSCs and NRCs design, their advantages, as well as theoretical problems and practical challenges of their realization. The influence of the NSCs and NRCs on the architectures of SDRs and CRs is also discussed.