Showing papers on "Root-raised-cosine filter published in 2015"

Improving Reliability of Particle Filter-Based Localization in Wireless Sensor Networks via Hybrid Particle/FIR Filtering

Abstract: The need for accurate, fast, and reliable indoor localization using wireless sensor networks (WSNs) has recently grown in diverse areas of industry Accurate localization in cluttered and noisy environments is commonly provided by means of a mathematical algorithm referred to as a state estimator or filter The particle filter (PF), which is the most commonly used filter in localization, suffers from the sample impoverishment problem under typical conditions of real-time localization based on WSNs This paper proposes a novel hybrid particle/finite impulse response (FIR) filtering algorithm for improving reliability of PF-based localization schemes under harsh conditions causing sample impoverishment The hybrid particle/FIR filter detects the PF failures and recovers the failed PF by resetting the PF using the output of an auxiliary FIR filter Combining the regularized particle filter (RPF) and the extended unbiased FIR (EFIR) filter, the hybrid RP/EFIR filter is constructed in this paper Through simulations, the hybrid RP/EFIR filter demonstrates its improved reliability and ability to recover the RPF from failures

The Guided Bilateral Filter: When the Joint/Cross Bilateral Filter Becomes Robust

Abstract: The bilateral filter and its variants, such as the joint/cross bilateral filter, are well-known edge-preserving image smoothing tools used in many applications. The reason of this success is its simple definition and the possibility of many adaptations. The bilateral filter is known to be related to robust estimation. This link is lost by the ad hoc introduction of the guide image in the joint/cross bilateral filter. We here propose a new way to derive the joint/cross bilateral filter as a particular case of a more generic filter, which we name the guided bilateral filter. This new filter is iterative, generic, inherits the robustness properties of the robust bilateral filter, and uses a guide image. The link with robust estimation allows us to relate the filter parameters with the statistics of input images. A scheme based on graduated nonconvexity is proposed, which allows converging to an interesting local minimum even when the cost function is nonconvex. With this scheme, the guided bilateral filter can handle non-Gaussian noise on the image to be filtered. A complementary scheme is also proposed to handle non-Gaussian noise on the guide image even if both are strongly correlated. This allows the guided bilateral filter to handle situations with more noise than the joint/cross bilateral filter can work with and leads to high peak signal-to-noise ratio values as shown experimentally.

Constant Time Median and Bilateral Filtering

Abstract: This paper formulates both the median filter and bilateral filter as a cost volume aggregation problem whose computational complexity is independent of the filter kernel size. Unlike most of the previous works, the proposed framework results in a general bilateral filter that can have arbitrary spatial$$^{1}$$1 and arbitrary range filter kernels. This bilateral filter takes about 3.5 s to exactly filter a one megapixel 8-bit grayscale image on a 3.2 GHz Intel Core i7 CPU. In practice, the intensity/range and spatial domain can be downsampled to improve the efficiency. This compression can maintain very high accuracy (e.g., 40 dB) but over $$100\times $$100? faster.

Double-talk detection for acoustic echo cancellation

Abstract: An acoustic echo canceller includes an adaptive filter and a double-talk detector. The adaptive filter includes a linear filter and a coefficient calculator. The linear filter has a transfer function that is controlled by a set of variable filter coefficients and that is configured to cancel an estimate of echo in a microphone signal to provide an output signal. The coefficient calculator is configured to update the set of variable filter coefficients based on a variable adaptation rate. The double-talk detector is configured to calculate changes in the energy of the variable filter coefficients (between updates of the coefficients). The acoustic echo canceller is configured to adjust the variable adaptation rate based on whether the energy of the variable filter coefficients is determined to be either oscillating or steadily changing (increasing or decreasing).

Efficient design of non-uniform cosine modulated filter banks for digital hearing aids

Abstract: In this paper, efficient and simple designs of non-uniform filter banks for digital hearing aid applications are proposed. Hearing aids should be individually tuned to satisfy the requirements of hearing impaired persons. Cosine modulated filter banks are one popular filter bank having simple design procedure with efficient implementation structure. In the proposed structure for hearing aid, non-uniform subbands are obtained using two methods. The first method is by merging the adjacent channels of a uniform filter bank and the second method, is by using transition filters between two filters with different bandwidths. The advantages of the proposed structure are simple design procedure, less implementation complexity, greater flexibility in tuning the subbands for various types of audiograms and improved performance in terms of matching error.

Global Color Sparseness and a Local Statistics Prior for Fast Bilateral Filtering

Abstract: The property of smoothing while preserving edges makes the bilateral filter a very popular image processing tool. However, its non-linear nature results in a computationally costly operation. Various works propose fast approximations to the bilateral filter. However, the majority does not generalize to vector input as is the case with color images. We propose a fast approximation to the bilateral filter for color images. The filter is based on two ideas. First, the number of colors, which occur in a single natural image, is limited. We exploit this color sparseness to rewrite the initial non-linear bilateral filter as a number of linear filter operations. Second, we impose a statistical prior to the image values that are locally present within the filter window. We show that this statistical prior leads to a closed-form solution of the bilateral filter. Finally, we combine both ideas into a single fast and accurate bilateral filter for color images. Experimental results show that our bilateral filter based on the local prior yields an extremely fast bilateral filter approximation, but with limited accuracy, which has potential application in real-time video filtering. Our bilateral filter, which combines color sparseness and local statistics, yields a fast and accurate bilateral filter approximation and obtains the state-of-the-art results.

Prototype Filter Design Approaches for Near Perfect Reconstruction Cosine Modulated Filter Banks - A Review

Abstract: Cosine Modulated Filter Banks (CMFB) are very popular among the different maximally decimated filter banks due to their design ease and simplicity in implementation and the property that all the coefficients of all the filters are real. All the analysis and synthesis filters are derived from one or two prototype filters. Hence, recently, the design of the prototype filter in a CMFB has become a subject of interest in the field of multirate signal processing. Perfect Reconstruction (PR) filter banks are those which can produce at the output, a weighted delayed version of the input. But in most of the applications a near perfect reconstruction (NPR) is sufficient. This can reduce the computational complexity. Different approaches developed for the efficient and optimal design of the prototype filter in a NPR orthogonal CMFB are studied, classified and summarized in this paper. In today's applications, less space and low power consumption are very essential. When the different design approaches for the design of the prototype filter in CMFB are compared, it is observed that the one using frequency response masking(FRM) and meta-heuristic optimization techniques gives better performance in terms of implementation complexity, which in turn can lead to reduced chip size and power consumption. It is hoped that this review will be highly beneficial to the researchers working in the area of multirate signal processing. At the end, we also propose some novel design approaches for the design of low complexity prototype filter using FRM technique.

Design of low complexity adjustable filter bank for personalized hearing aid solutions

Abstract: The emerging demand for personalized hearing aids requires the filter bank of a hearing aid system to be capable of decomposing the sound waves in accordance with the characteristic of the patient’s hearing loss. In this paper, an efficient adjustable filter bank is proposed to achieve this goal. By careful design, the number of the subbands as well as the location of the subbands can be easily adjusted by changing a 4-bit control signal. The proposed filter bank has extremely low complexity due to the adoption of fractional interpolation and the technique of symmetric and complementary filters. Only one prototype filter is needed for each of the stages, the multiple passbands generation stage and masking stage. We show, by means of examples, that the proposed filter bank can meet different needs of hearing loss cases with acceptable delay.

Removal of noise from electrocardiogram using digital FIR and IIR filters with various methods

Abstract: Electrocardiogram (ECG) is a type of measuring the electrical activities of heart. Each section of ECG is necessary for the diagnosis of various cardiac problems. But the amplitude and time period of ECG signal is generally corrupted by various noises. After an analog ECG signal is transformed into digital format, appropriate digital filter can be utilized to repress the various kinds of noise like Baseline Wander, Power line Interference, High -frequency Noise, Physiological Artifacts etc., depends on their specifications. In generic two types of method can be classified in this paper; FIR filters like Rectangular, Hann, Blackman, Hamming and Kaiser window techniques and IIR filters like Butterworth, Chebyshev I, Chebyshev II and Elliptic filters are also prospected to reduce artifacts in ECG signal. The results are collected from different orders for FIR filter as 56, 300, 450, and 600 and for IIR filter as 1, 2, and 3. The signals taken from the MIT-BIH data base which contains the normal and abnormal waveforms. The work has been implemented in MATLAB FDA Tool. The results are obtained using different window based FIR filters, IIR filter with different approximation methods and their respective waveforms are shown. In addition, power spectrum density, signal to noise ratio (SNR) and means square error (MSE) of both noisy and filtered ECG signals are calculated. We observed that Digital FIR filter with Kaiser Window in order 56 shows high performance as compared to the other windowing techniques and Digital IIR filter approximation methods.

Filtering strength determination method, moving picture coding method and moving picture decoding method

Abstract: A moving picture coding apparatus includes an inter-pixel filter having filters for filtering decoded image data so as to remove block distortion which is high frequency noise around block boundaries. The inter-pixel filter includes filters having different filtering strengths. The coding apparatus also includes a filter processing control unit for determining a filtering strength of the inter-pixel filter.

Efficient IIR notch filter design using Minimax optimization for 50Hz noise suppression in ECG

Abstract: Quality of Electrocardiogram (ECG) signal is very important for patient health diagnostics. ECG consists of frequencies ranging from 0.01 – 300Hz, while being a very small magnitude signal. ECG usually gets corrupted by the 50Hz power line interference (PLI). Thus, filtering out the PLI is a necessary task for the right diagnostics. In this work a second order efficient digital infinite impulse response (IIR) notch filter is designed to suppress PLI. For this task, Minimax optimization technique has been utilized to minimize the root mean square error (RMSE) defined as the difference of magnitude response of ideal and proposed IIR notch filter designs. The optimization process was able to set pole position close to the unit circle and the pass band gain is so chosen so as to have symmetrical performance resulting in attenuation at 50Hz notch of −260dB, a gain of 0.995 and −3dB bandwidth of 2.375Hz. The performance of the designed filter has been investigated theoretically as well as validated experimentally on a contaminated ECG signal and a pure sinusoid signal of 50Hz on field programmable gate array (FPGA) in the LabVIEW environment. In simulation, a power spectral density (PSD) of −30dB has been achieved for the simulated ECG signal originally having 4dB PSD at 50Hz and an attenuation of −30dB in PSD has been obtained for a pure sinusoidal signal. The designed filter was also implemented on FPGA a PSD of −26dB was obtained for ECG at 50Hz and −26dB for sinusoidal signal, respectively. Based on these studies it can be concluded that the proposed Minimax based notch filter design is an efficient one.

Improved Goldstein SAR Interferogram Filter Based on Adaptive-Neighborhood Technique

Abstract: The Goldstein filter is one of the most commonly used filters for synthetic aperture radar (SAR) interferograms. The level of noise after filtering is controlled by a filter parameter, "alpha," the value of which is determined by pixels within the moving window. However, when there exist different features within a single filter window, especially along the border, the value of alpha as estimated from the pixels within the window can be inaccurate and this may result in blurred borders in filtered interferograms. This letter proposes a modified Goldstein filter based on the adaptive-neighborhood technique. The idea of this method is to filter each pixel of the interferogram within an adjusted filter patch. In this adjusted patch, the adaptive-neighborhood pixels retain the original phase values while the "background" pixels are replaced by the mean value of adaptive-neighborhood pixels. Then, the Fourier transform of the complex phase is applied to this adjusted filter patch. The difficulty of estimating the noise level near the borders of different features can be decreased using this new filtering method. The quantitative results from real data show that this newly developed method could reduce the phase noise efficiently while also outperforming the Goldstein, Baran and empirical mode decomposition (EMD) filters by preserving the edges in interferograms.

The class of generalized hampel filters

Abstract: The standard median filter has only one tuning parameter — the width of the moving window on which it is based — and this has led to the development of a number of extremely useful extensons, including the recursive median filter, weighted median filters, and recursive weighted median filters. The Hampel filter is a member of the class of decision filters that, as we note here, may be viewed as another generalization of the median filter. This paper exploits this relationship, defining and briefly exploring the class of generalized Hampel filters, obtained by applying the median filter extensions listed above.

Global navigation satellite system interference tracking and mitigation based on an adaptive fading Kalman filter

Abstract: A time-domain signal tracking and mitigation algorithm is proposed to estimate the frequency of interference in a global navigation satellite system (GNSS) so as to classify different types of interference and to mitigate interference. The frequency of GNSS interference can be obtained using the properties of the trigonometric functions of received signal samples, but these values contain numerous errors caused by measurement noise and frequency changes associated with the interference. To reduce these errors, an adaptive fading Kalman filter is applied in the proposed algorithm. Furthermore, a low-pass differentiator and a pattern enhancement algorithm are used to estimate the sweep period of chirp-type interference, which is used to reset the filter parameter for estimating the frequency of the interference accurately. By estimating the sweep period, the interference identification logic is designed to select the proper system model of the Kalman filter. Finally, in order to mitigate the interference, the denoised frequency from the filter is used to design a notch filter which eliminates the interference in the received signal. The frequency tracking performance of the proposed algorithm is verified to compare with conventional algorithms and the mitigation performance of the proposed algorithm is evaluated by means of Monte Carlo simulations.

A survey on switching median filters for impulse noise removal

Abstract: Impulse noise is one of the major problems in digital image processing. Impulse noise can corrupt the images and also damage the fine details in the image. Different filtering algorithms are applied on images to remove the impulse noise that are present in image during acquisition. Corrupted pixels in image take either the minimum or maximum gray level. Image denoising filters are based on the median filter. Survey on Variants of median filters like weighted median filter, adaptive switching median filter, triststate switching median filter, modified decision based unsymmentrictristate median filter are discussed in his paper.

Epoch Extraction Using Zero Band Filtering from Speech Signal

Abstract: Zero frequency filter (ZFF) is a marginally stable infinite impulse response resonant filter at 0 Hz that is used to extract the epoch locations reliably from speech signals. However, the output of such an ideal resonator is an exponentially increasing/decreasing function of time. The trend is removed from the filtered output by subtracting the average over 1---2 pitch periods to obtain zero frequency filtered signal. Alternatively in this paper, a bounded input bounded output stable realization of ZFF is proposed for epoch extraction, where the output of such a filter is not an increasing/decreasing function of time. The advantages of using such a stable filter is that the filter output is bounded and has no precision related problem associated with the output for lengthy speech files, also, the method does not require remove trend procedure that needs initial pitch estimation. The proposed approach is evaluated using CMU-Arctic database for clean and degraded conditions. Furthermore, the method is also validated in cases of singing voice and emotional speech to demonstrate the robustness for varying pitch scenarios. The proposed method is found to be robust for wide range of chosen parameters.

A Novel Approach to Design Low-Cost Two-Stage Frequency-Response Masking Filters

Abstract: The multistage frequency-response masking (FRM) technique is widely used to reduce the complexity of a filter when the transition bandwidth is extremely small In this brief, a real generalized two-stage FRM filter without any constraint on the subfilters or the interpolation factors was proposed New principles and equations were deduced to determine the design parameters The subfilters were then jointly optimized using nonlinear optimization Experiential results show that when the proposed algorithm obtains different solutions with the conventional algorithm, the solution of the proposed approach is better with less number of filter coefficients and sometimes with lower delay as well than the conventional two-stage FRM, which can lead to a reduced hardware cost in applications

Recursive Digital Filters With Tunable Lag and Lead Characteristics for Proportional-Differential Control

Abstract: Regression analysis using orthogonal polynomials in the time domain is used to derive a digital filter with an infinite impulse response that satisfies maximally flat design constraints near dc. The low-frequency phase, and high-frequency gain, may be adjusted for lead or lag compensation of plant dynamics. Simulated design examples are used to show how the compensating filter may be intuitively tuned for the desired closed-loop response. It is shown that the second-order instantiation of the compensating filter reduces to a proportional–differential plus filter controller, with improved noise attenuation; closed-form expressions for the filter coefficients, as a function of two design parameters, are provided.

An event driven technique for filtering computational complexity reduction

Abstract: This work is a contribution to enhance the signal processing chain required in mobile systems like satellites, cell phones, biomedical implants remote motors, etc. The system is powered by a battery therefore it must be power efficient. Filtering is a basic operation, almost required in every signal processing system. The classical filtering is time-invariant, the sampling frequency and the filter order remains unique. Therefore it can render a useless increase of the processing activity, especially in the case of sporadic signals. In this context an adaptive rate filtering technique, based on an event driven sampling is devised. It adapts the sampling frequency and the filter order by analysing the input signal characteristics. It correlates the processing activity to the signal variations. The computational complexity and the output quality of the proposed technique are compared to the classical one for a speech signal. Results show a drastic computational gain, of the proposed technique compared to the classical one, along with a comparable output quality.

The multinotch, part I: A low latency, high numerical fidelity filter for mechatronic control systems

Abstract: Control of lightly damped mechatronic systems is often accomplished in practice with a PID-like controller in series with a filter to limit the effects of high frequency resonances. The high frequency filtering is often limited by an inability to precisely match multiple lightly damped resonances with a digital filter, and by the extra computational delay of the such filters. The multinotch is a filter topology that addresses these issues, allowing for precise matches to many lightly damped resonances and anti-resonances, while maintaining a small and fixed computational delay [1]. Coefficient adjustments for higher precision when the resonant dynamics to be filtered span a large frequency range are described in [2].

Combined line-of-sight error and angular position to generate feedforward control for a charge-coupled device-based tracking loop

Abstract: A feedforward control based on data fusion is proposed to enhance closed-loop performance. The target trajectory as the observed value of a Kalman filter is recovered by synthesizing line-of-sight error and angular position from the encoder. A Kalman filter based on a Singer acceleration model is employed to estimate the target velocity. In this control scheme, the control stability is influenced by the bandwidth of the Kalman filter and time misalignment. The transfer function of the Kalman filter in the frequency domain is built for analyzing the closed loop stability, which shows that the Kalman filter is the major factor that affects the control stability. The feedforward control proposed here is verified through simulations and experiments. (C) 2015 Society of Photo-Optical Instrumentation Engineers (SPIE)

A generalized design framework for IIR digital multiple notch filters

Abstract: Digital multiple notch filters are used in a variety of applications to remove or suppress multiple sinusoidal or narrow-band interference in digital signals. In this paper, we propose an all-pass filter-based design framework for infinite impulse response (IIR) multiple notch filters. Our approach aims to overcome the limitations of former techniques through greater design capacity and performance. The proposed framework has versatility and enables the tailored use of design constraints thus providing a family of possible multiple notch filter design methods. The design performance and practicality of the proposed framework are verified empirically by a series of experimental results and different applications.