scispace - formally typeset
Search or ask a question

Showing papers on "Linear phase published in 2013"


Journal ArticleDOI
Liang Wang1, Qirong Jiang1, Lucheng Hong1, Chunpeng Zhang1, Yingdong Wei1 
TL;DR: In this paper, a three-phase software phase-locked loop (PLL) is proposed to operate fast and accurately in unbalanced, polluted, and frequency deviating circumstances, which consists of a frequency detector and an initial phase angle detector.
Abstract: This paper proposes a new three-phase software phase-locked loop (PLL) that operates fast and accurately in unbalanced, polluted, and frequency deviating circumstances. The proposed PLL consists of a frequency detector and an initial phase angle detector. In the synchronous reference frame, the initial phase angle detector tracks a ramp phase angle, which is generated from frequency deviation of the inputs, with steady error. This detection error is utilized to estimate the actual grid frequency. Frequency adaptive moving average filters (MAFs) are applied in this new PLL to eliminate noises, harmonics, and negative sequence components. In this paper, the effect of discrete sampling on the MAFs is analyzed, and a linear interpolation is employed to enhance the performances of the MAFs. The stability of the proposed PLL is also analyzed, a sufficient stability condition is identified, and the design procedures of the control parameters are also presented. Simulations and experiments verify the performances of the novel PLL.

101 citations


Journal ArticleDOI
TL;DR: The design results included in the paper clearly show the improvement of the proposed PSO technique over earlier reported results.
Abstract: In this paper, a new particle swarm optimization (PSO) based method is proposed for the design of a two-channel linear phase quadrature mirror filter (QMF) bank in frequency domain. The origional particle swarm optimization technique is modified by introducing the concept of Scout Bee from Artificial Bee Colony (ABC) technique for designing a low pass prototype filter having ideal filter characteristics in the passband and stopband regions, and its magnitude response at quadrature frequency is 0.707. The design problem is formulated as a linear combination of passband error and residual stop band energy of the low pass filter, and the square error of the overall transfer function of the QMF bank at the quadrature frequency π/2, in the transition band. The design results included in the paper clearly show the improvement of the proposed PSO technique over earlier reported results.

61 citations


Proceedings ArticleDOI
26 May 2013
TL;DR: This paper proposes a novel blind compensation of sampling frequency mismatch for asynchronous microphone array by assuming the sources are motionless and stationary and the maximum likelihood estimation is obtained effectively by a golden section search.
Abstract: This paper proposes a novel blind compensation of sampling frequency mismatch for asynchronous microphone array. Digital signals simultaneously observed by different recording devices have drift of the time differences between the observation channels because of the sampling frequency mismatch among the devices. Based on the model that such the time difference is constant within each time frame, but varies proportional to the time frame index, the effect of the sampling frequency mismatch can be compensated in the short-time Fourier transform domain by the linear phase shift. By assuming the sources are motionless and stationary, a likelihood of the sampling frequency mismatch is formulated. The maximum likelihood estimation is obtained effectively by a golden section search.

54 citations


Patent
11 Mar 2013
TL;DR: In this article, a disk drive is disclosed comprising a head actuated over a disk comprising a sector including a periodic pattern and sector data, and the sector is read with the head to generate a read signal which is sampled at a sampling frequency with a signal sampler to generate signal samples.
Abstract: A disk drive is disclosed comprising a head actuated over a disk comprising a sector including a periodic pattern and sector data. The sector is read with the head to generate a read signal which is sampled at a sampling frequency with a signal sampler to generate signal samples. The signal samples representing the periodic pattern are processed to measure a frequency induced phase error based on kδ where k represents a signal sample index and δ is a fraction of 2π. The signal samples representing the sector data are processed to generate a data phase error. The data phase error is adjusted in response to the frequency induced phase error to generate an adjusted data phase error, and the signal sampler is controlled in response to the adjusted data phase error.

42 citations


Journal ArticleDOI
TL;DR: Experimental results show that filters designed using the proposed method have much lower group-delay deviation for the same passband ripple and stopband attenuation when compared with corresponding filters designed with several state-of-the-art competing methods.
Abstract: A new optimization method for the design of nearly linear-phase IIR digital filters that satisfy prescribed specifications is proposed. The group-delay deviation is minimized under the constraint that the passband ripple and stopband attenuation are within the prescribed specifications and either a prescribed or an optimized group delay can be achieved. By representing the filter in terms of a cascade of second-order sections, a non-restrictive stability constraint characterized by a set of linear inequality constraints can be incorporated in the optimization algorithm. An additional feature of the method, which is very useful in certain applications, is that it provides the capability of constraining the maximum gain in transition bands to be below a prescribed level. Experimental results show that filters designed using the proposed method have much lower group-delay deviation for the same passband ripple and stopband attenuation when compared with corresponding filters designed with several state-of-the-art competing methods.

42 citations


Journal ArticleDOI
TL;DR: A new two-way time synchronization method for orthogonal frequency division multiplexing (OFDM)-based wireless systems that relies on the time-reversal technique to remove the effect of the channel phase after the signal round-trip, named TR-TS.
Abstract: In this paper, we propose a new two-way time synchronization (TS) method for orthogonal frequency division multiplexing (OFDM)-based wireless systems. It relies on the time-reversal (TR) technique to remove the effect of the channel phase after the signal round-trip. Thus, the method is named TR-TS. TR technique yields a linear phase rotation across subcarriers, regardless whether the channel is minimum, maximum, or mixed phase. This phase is proportional to the difference of the OFDM symbol-timing synchronization errors at the two receivers. Thus, the clock-offset between the radios is determined using the local reception times and the estimated linear phase resulting from the TR technique. A reliable low-complexity algorithm called fast Fourier transform-weighted least-squares is proposed to estimate the linear phase slope, and its mean-square error is compared with the Cramer-Rao Lower Bound (CRLB). The results show that the proposed algorithm attains the CRLB at low SNR, even when a single OFDM symbol is used. The OFDM packets are time-stamped at the medium access control layer, which allows eliminating errors caused by different and varying delays in the internal lower level processing of the data. Hence, an accurate estimate of the clock-offset is obtained. The impact of various nonidealities on the proposed algorithm is also studied. In addition, we propose a ranging method that employs the novel TS method and a first-path delay estimation technique. The performance of the proposed methods is studied in simulations, as well as using real-world measured channels. The results show that the proposed methods can be successfully applied in low to moderate mobility scenarios such as indoors despite harsh multipath, since they rely on channel reciprocity.

39 citations


Journal ArticleDOI
TL;DR: A comparison of simulation results reveals the optimization efficacy of the OHS over the other optimization techniques for the solution of the multimodal, nondifferentiable, nonlinear, and constrained FIR filter design problems.
Abstract: In this paper, opposition-based harmony search has been applied for the optimal design of linear phase FIR filters. RGA, PSO, and DE have also been adopted for the sake of comparison. The original harmony search algorithm is chosen as the parent one, and opposition-based approach is applied. During the initialization, randomly generated population of solutions is chosen, opposite solutions are also considered, and the fitter one is selected as a priori guess. In harmony memory, each such solution passes through memory consideration rule, pitch adjustment rule, and then opposition-based reinitialization generation jumping, which gives the optimum result corresponding to the least error fitness in multidimensional search space of FIR filter design. Incorporation of different control parameters in the basic HS algorithm results in the balancing of exploration and exploitation of search space. Low pass, high pass, band pass, and band stop FIR filters are designed with the proposed OHS and other aforementioned algorithms individually for comparative optimization performance. A comparison of simulation results reveals the optimization efficacy of the OHS over the other optimization techniques for the solution of the multimodal, nondifferentiable, nonlinear, and constrained FIR filter design problems.

36 citations


Journal ArticleDOI
TL;DR: In this work, a novel genetic algorithm (GA) is proposed for the design of multiplierless linear phase finite impulse response (FIR) filters, and significantly outperforms existing algorithms dealing with the similar problems in terms of design time and hardware cost.
Abstract: In this work, a novel genetic algorithm (GA) is proposed for the design of multiplierless linear phase finite impulse response (FIR) filters. The filters under consideration are of high order and wide coefficient wordlength. Both the single-stage and cascade form are considered. In a practical filter design problem, when the filter specification is stringent, requiring high filter order and wide coefficient wordlength, GAs often fail to find feasible solutions, because the discrete search space thus constructed is huge and the majority of the solution candidates therein can not meet the specification. In the proposed GA, the discrete search space is partitioned into smaller ones. Each small space is constructed surrounding a base discrete coefficient set which is obtained by a proposed greedy algorithm. The partition of the search space increases the chances for the GA to find feasible solutions, but does not sacrifice the coverage of the search. The proposed GA applies to the design of single-stage filters. When a cascade form filter is designed, for each single-stage filter meeting the filter specification generated during the course of GA, an integer polynomial factorization is applied. Design examples show that the proposed GA significantly outperforms existing algorithms dealing with the similar problems in terms of design time, and the hardware cost is saved in most cases.

33 citations


Journal ArticleDOI
TL;DR: The fractional derivative constraint is used to improve the design accuracy of 2-D FIR filter and the performance comparison with integer derivative method is demonstrated.

33 citations


Journal ArticleDOI
TL;DR: It is found that the clutter calibration algorithm performs best for statistically homogeneous scenes but that the contrast-calibration algorithms perform better with scenes with larger contrast ratios.
Abstract: A contrast-based phase calibration algorithm for digital beamforming remote sensing radars using three contrast metrics is presented. The algorithm corrects time-varying antenna array phase errors that defocus digital beamforming remote sensing radar imagery. Amplitude errors are treated by equalizing the received powers in all elements. As such, the algorithm does not produce an absolute (or radiometric) calibration vector for the array. The performance of the algorithm is studied using a combination of simulated and real radar data under various conditions and is compared with a clutter-based calibration algorithm. An analytical proof showing that maximizing the expected value of the 4-norm metric is equivalent to phase-calibrating the image, except for a linear phase offset, is provided. We find that the clutter calibration algorithm performs best for statistically homogeneous scenes but that the contrast-calibration algorithms perform better with scenes with larger contrast ratios.

32 citations


Journal ArticleDOI
TL;DR: A new procedure for design of linear phase IIR notch filters using parallel connection of two IIR all-pass filters with approximately linear phase exhibits fast convergence and easy initial values determination.

Journal ArticleDOI
TL;DR: In this paper, a single-pole group delay equalizer was also given to flatten a nonlinear phase quasi-elliptic filter's phase, which solved the tuning difficulty of the HTSC linear phase filter.
Abstract: A variety of miniature half-wave multizigzag resonators was given, and an eight-pole high-temperature superconductor (HTSC) quasi-elliptic filter with no cross-line was developed with these resonators. In this paper, a novel single-pole group delay equalizer was also given to flatten a nonlinear phase quasi-elliptic filter's phase, which solves the tuning difficulty of the HTSC linear phase filter. Moreover, it cannot only ensure that the finished HTSC filter has a good linear phase characteristic but also solve the integration of the linear phase filter with external equalization. Finally, on YBCO/LaAlO3/YBCO substrate with the dimension of 31.6 mm × 29.2 mm, a miniature eight-pole HTSC quasi-elliptic linear phase filter using external equalization is developed, where the center frequency is 2250.84 MHz, the bandwidth is 14 MHz, the best insertion loss in passband is 2.09 dB, and the return loss is better than 18.97 dB, as per measurements. In passband, the measured group delay response has a good agreement with the simulated response, and its variation is less than 50 ns over 78.5% of the filter bandwidth.

Journal ArticleDOI
TL;DR: This letter shows that a PLL with fixed gain achieves a performance that is close to the optimal one provided that the phase detector is optimized.
Abstract: Data-aided carrier recovery based on phase-lock loop (PLL) is a popular scheme for tracking the phase of an incoming carrier affected by phase noise. Optimum tracking is achieved by a Kalman filter, that, with multilevel quadrature amplitude modulation (QAM), is implemented by a PLL with variable loop gain. This letter shows that a PLL with fixed gain achieves a performance that is close to the optimal one provided that the phase detector is optimized. Monte Carlo simulations for the mean-square phase error and the symbol error rate of a 256-QAM constellation are provided to validate the analysis.

Proceedings ArticleDOI
TL;DR: A method to convert high-frequency waveforms into low- frequencies waveforms based on a linear phase approximation at the low frequency end and the resulted low- frequency waveform can benefit the convergence in full waveform inversion.
Abstract: Summary In full waveform inversion, it is preferred to use the multiscale approach which starts iteration from low-frequency data to determine the large-scale heterogeneities first. This approach tends to provide better convergence when the initial model is drastically biased from the true velocity model. However, the combined passband of the source and geophone usually generates data lacks the low-frequency information preferred by the full waveform inversion. On the other hand, the travel time data are often used to generate low-resolution initial models. The low-frequency waveform and the travel-time information share some common properties. The successful use of travel time data implies that the seismic data are weakly dispersed at least at low frequencies. This property can be used to extrapolate the phase information in the data towards lower frequencies. Motivated by this idea, we propose a method to convert high-frequency waveforms into low-frequency waveforms based on a linear phase approximation at the low frequency end. The resulted low-frequency waveform can benefit the convergence in full waveform inversion. To validate the proposed method, we compare waveforms converted from high-frequency traces with those actually generated using low-frequency sources. We successfully tested the converted data set in waveform inversion. The inversion using the high-frequency converted lowfrequency seismic data converge better than those directly using the original high-frequency data.

Journal ArticleDOI
TL;DR: An efficient approximation to the nonlinear phase diversity (PD) method for wavefront reconstruction and correction from intensity measurements with potential of being used in real-time applications is proposed.
Abstract: We propose an efficient approximation to the nonlinear phase diversity (PD) method for wavefront reconstruction and correction from intensity measurements with potential of being used in real-time applications. The new iterative linear phase diversity (ILPD) method assumes that the residual phase aberration is small and makes use of a first-order Taylor expansion of the point spread function (PSF), which allows for arbitrary (large) diversities in order to optimize the phase retrieval. For static disturbances, at each step, the residual phase aberration is estimated based on one defocused image by solving a linear least squares problem, and compensated for with a deformable mirror. Due to the fact that the linear approximation does not have to be updated with each correction step, the computational complexity of the method is reduced to that of a matrix-vector multiplication. The convergence of the ILPD correction steps has been investigated and numerically verified. The comparative study that we make demonstrates the improved performance in computational time with no decrease in accuracy with respect to existing methods that also linearize the PSF.

Journal ArticleDOI
TL;DR: A fast phase retrieval algorithm that is suitable for real-time applications such as adaptive optics and is valid for low-NA focused field is developed by linearising the pupil function in the approximation of small aberrations.
Abstract: We developed a fast phase retrieval algorithm that is suitable for real-time applications such as adaptive optics. The phase retrieval model is developed by linearising the pupil function in the approximation of small aberrations and is valid for low-NA focused field. The linear model in conjunction with a particular choice for the position of the single out-of-focus measurement plane and an efficient control algorithm, significantly reduces the computation time for phase retrieval. The experimental results demonstrate the validity of the described approach for fast correction of aberrations.

Journal ArticleDOI
TL;DR: The results demonstrate that the proposed problem solving approach blended with the use of the spiral optimization technique produced filters which fulfill the desired characteristics and are of practical use.
Abstract: The multiobjective design of digital filters using spiral optimization technique is considered in this paper. This new optimization tool is a metaheuristic technique inspired by the dynamics of spirals. It is characterized by its robustness, immunity to local optima trapping, relative fast convergence and ease of implementation. The objectives of filter design include matching some desired frequency response while having minimum linear phase; hence, reducing the time response. The results demonstrate that the proposed problem solving approach blended with the use of the spiral optimization technique produced filters which fulfill the desired characteristics and are of practical use.

Journal ArticleDOI
TL;DR: The Maxwell (sine)-Helmholtz (cosine) approach has proven successful for a horizontal phase gradient coil and a similar approach may be useful for other phase-gradient coil designs.

Journal ArticleDOI
TL;DR: It is shown for the first time that, in a coherent communication system that employs a phase-shift-keying signal and Raman amplification, besides the pump relative intensity noise, the signal's phase will also be affected by pump RIN through the pump-signal cross-phase modulation.
Abstract: We show for the first time, to the best of our knowledge, that, in a coherent communication system that employs a phase-shift-keying signal and Raman amplification, besides the pump relative intensity noise (RIN) transfer to the amplitude, the signal's phase will also be affected by pump RIN through the pump-signal cross-phase modulation. Although the average pump power induced linear phase change can be compensated for by the phase-correction algorithm, a relative phase noise (RPN) parameter has been found to characterize pump RIN induced stochastic phase noise. This extra phase noise brings non-negligible system impairments in terms of the Q-factor penalty. The calculation shows that copumping leads to much more stringent requirements to pump RIN, and relatively larger fiber dispersion helps to suppress the RPN induced impairment. A higher-order phase-shift keying (PSK) signal is less tolerant to noise than a lower-order PSK.

Patent
15 Mar 2013
TL;DR: In this paper, a digital frequency synthesizer provides absolute phase lock and shorter settling time through the use of a digital filter with a phase and frequency path and control logic control disables the frequency path during the frequency acquisition and sets a wide bandwidth.
Abstract: A digital frequency synthesizer provides absolute phase lock and shorter settling time through the use of a digital filter with a phase and frequency path. Control logic control disables the frequency path during the frequency acquisition and sets a wide bandwidth. After frequency acquisition, a counter with digital phase information is reset using the input clock signal to bring the output phase closer to lock with the input signal and the control logic enables the phase path in the digital loop filter to achieve phase lock with a narrower bandwidth than the initial bandwidth.

Journal Article
TL;DR: A modular FPGA-based 16 channel digital beamforming with embedded DSP for ultrasound imaging is presented and physical array elements shown a good results in linear phase array reconstruction (steering) than virtual array elements, because the active elements number (Aperture) is less than in physicalarray elements.
Abstract: Ultrasound imaging is an efficient, noninvasive, method for med ical diagnosis. A commonly used approach to image acquisition in ultrasound system is digital beamforming. Digital beamforming, as applied to the medical ultrasound, is defined as phase align ment and summation of signals that are generated fro m a co mmon source, by received at different times by a mu lti-elements ultrasound transducer. In this paper first: we tested all signal processing methodologies for digital beamforming which included: the effect of over samp ling techniques, single trans mit focusing and their limitations, the apodization technique and its effect to reduce the sidelobes, the analytical envelope detection using digital finite impulse response (FIR) filter appro ximations for the Hilbert transformat ion and how to co mpress the dynamic range to achieve the desired dynamic range for display (8 bits). Here the image was reconstructed using physical array elements and virtual array elements for linear and phase array probe. The results shown that virtual array elements were given well results in linear array image reconstruction than physical array elements, because it provides additional number of lines. Ho wever, physical array elements shown a good results in linear phase array reconstruction (steering) than virtual array elements, because the active elements number (Aperture) is less than in physical array elements. We checked the quality of the image using quantitative entropy. Second: a modular FPGA-based 16 channel digital u ltrasound beamforming with embedded DSP for ultrasound imaging is presented. The system is imp lemented in Virtex-5 FPGA (Xilin x, Inc.). The system consists of: tw o 8 channels block, the DSP wh ich co mposed of the FIR Hilbert filter bl ock to obtain the quadrature components, the fractional delay filter block (in-phase filter) to co mpensate the delay when we were used a high FIR order, and the envelope detection block to compute the envelope of the in-phase and quadrature components. The Hilbert filter is imp lemented in the form whereby the zero tap coefficients were not computed and therefore an order L filter used only L/2 mu ltiplications. This reduced the computational time by a half. Fro m the implementation result the total estimated power consumption equals 4732.87 mW and the device utilization was acceptable. It is possible for the system to accept other devices for further processing. Also it is possible to build 16-,32-, and 64-channel beamformer. The hardware architecture of the design provided flexib ility for beamforming.

Proceedings ArticleDOI
01 Oct 2013
TL;DR: This paper proposes a blind synchronization of ad-hoc microphone array in the short-time Fourier transform (STFT) domain with the optimized frame analysis centered at non-integer discrete time and shows that the drift caused by sampling frequency mismatch of asynchronous observation channels can be disregarded in a short interval.
Abstract: This paper proposes a blind synchronization of ad-hoc microphone array in the short-time Fourier transform (STFT) domain with the optimized frame analysis centered at non-integer discrete time. We show that the drift caused by sampling frequency mismatch of asynchronous observation channels can be disregarded in a short interval. Utilizing this property, the sampling frequency mismatch and the recording start offset are estimated roughly by finding two pairs of the short intervals corresponding to the same continuous time. Using the estimate, STFT analysis is synchronized roughly between channels with optimized frame central. Since the optimized frame central is generally non-integer, we approximate the frame analysis by the linear phase filtering of the frame centered at the nearest integer sample. Maximum likelihood estimation refines the compensation of sampling frequency mismatch.

Journal ArticleDOI
TL;DR: Some efficient algorithms exploiting the coefficients' matrix nature for the constrained least-squares (CLS) and minimax designs of quadrantally symmetric 2-D linear-phase FIR filters, both of which can be formulated as an optimization problem or converted into a sequence of subproblems with a positive-definite quadratic cost.
Abstract: The impulse response coefficients of a two-dimensional (2-D) finite impulse response (FIR) filter are in a matrix form in nature. Conventional optimal design algorithms rearrange the filter's coefficient matrix into a vector and then solve for the coefficient vector using design algorithms for one-dimensional (1-D) FIR filters. Some recent design algorithms have exploited the matrix nature of the 2-D filter's coefficients but not incorporated with any constraints, and thus are not applicable to the design of 2-D filters with explicit magnitude constraints. In this paper, we develop some efficient algorithms exploiting the coefficients' matrix nature for the constrained least-squares (CLS) and minimax designs of quadrantally symmetric 2-D linear-phase FIR filters, both of which can be formulated as an optimization problem or converted into a sequence of subproblems with a positive-definite quadratic cost and a finite number of linear constraints expressed in terms of the filter's coefficient matrix. Design examples and comparisons with several existing algorithms demonstrate the effectiveness and efficiency of the proposed algorithms.

Proceedings ArticleDOI
23 Jun 2013
TL;DR: It is shown that the presence of bilateral symmetry in the 3-D shape is equivalent to a linear phase structure in the corresponding spherical harmonic coefficients, and algorithms for estimating the orientation of the symmetry plane are provided.
Abstract: We show that bilateral symmetry plane estimation for three-dimensional (3-D) shapes may be carried out accurately, and efficiently, in the spherical harmonic domain. Our methods are valuable for applications where spherical harmonic expansion is already employed, such as 3-D shape registration, morphometry, and retrieval. We show that the presence of bilateral symmetry in the 3-D shape is equivalent to a linear phase structure in the corresponding spherical harmonic coefficients, and provide algorithms for estimating the orientation of the symmetry plane. The benefit of using spherical harmonic phase is that symmetry estimation reduces to matching a compact set of descriptors, without the need to solve a correspondence problem. Our methods work on point clouds as well as large-scale mesh models of 3-D shapes.

Proceedings ArticleDOI
01 Aug 2013
TL;DR: The proposed parallel FIR structures not only use fast convolution algorithm to reduce the number of subfilter, but also exploit the symmetric (or antisymmetric) coefficients of linear-phase FIR filter to reduce half theNumber of multiplications in subfilter section at the expense of additional adders in preprocessing and postprocessing blocks.
Abstract: Based on fast convolution algorithm, this paper proposes improved parallel FIR filter structures for linear-phase FIR filters where the number of taps is a multiple of parallelism. The proposed parallel FIR structures not only use fast convolution algorithm to reduce the number of subfilter, but also exploit the symmetric (or antisymmetric) coefficients of linear-phase FIR filter to reduce half the number of multiplications in subfilter section at the expense of additional adders in preprocessing and postprocessing blocks. The proposed parallel FIR structures save a large amount of hardware cost for symmetric (or antisymmetric) coefficients from the reported FFA parallel FIR filter structures, especially when the length of the filter is large, e.g., the proposed 4-parallel FIR filter structure has eight subfilter blocks in total and four subfilter blocks contain symmetric coefficients, whereas the improved FFA structure has nine subfilter blocks in total and four contain symmetric coefficients. Specifically, for a 4-parallel 576-tap filter, the proposed design saves 144 multipliers (14.3%), 135 adders (10.2%) and 143 delay elements (11.1%).

Journal ArticleDOI
TL;DR: The relationship between the pull-in range of a linear phase-locked loop (PLL) and the current mismatch of the charge pump that controls the frequency of the oscillator in the PLL is shown.
Abstract: In this paper, we show the relationship between the pull-in range of a linear phase-locked loop (PLL) and the current mismatch of the charge pump that controls the frequency of the oscillator in the PLL. We evaluate the pull-in range of the PLL based on a nonlinear behavioral model of the pull-in process with three types of phase detectors. We introduce the mismatch error to the charge pump current in the PLL and study the impact of this error on the pull-in range. We also apply the mismatch error to a nonlinear differential equation that describes the loop dynamics of the PLL and calculate the pull-in range under this mismatch condition. We validate the limitations of the pull-in range due to the current mismatch by a numerical simulation with MATLAB.

Journal ArticleDOI
TL;DR: A new implementation of microwave-photonic filters (MPFs) based on tunable optical delay lines is proposed and demonstrated, based on mapping of the spectral components of an incoming waveform onto the time domain, the application of linearly-varying temporal phase offsets, and an inverse mapping back to the frequency domain.
Abstract: A new implementation of microwave-photonic filters (MPFs) based on tunable optical delay lines is proposed and demonstrated. The variable delay is based on mapping of the spectral components of an incoming waveform onto the time domain, the application of linearly-varying temporal phase offsets, and an inverse mapping back to the frequency domain. The linear phase correction is equivalent to a frequency offset, and realized though suppressed-carrier single-sideband modulation by a radio-frequency sine wave. The variable delay element, controlled by the selected frequency, is used in one arm of a two-tap MPF. In a proof-of-concept experiment, the free spectral range (FSR) of the MPF was varied by over a factor of four: between 1.2 GHz and 5.3 GHz.

Journal ArticleDOI
TL;DR: Experimental results show that beamformers designed using the proposed methods have much smaller passband group-delay deviation for similar passband ripple and stopband attenuation than a modified version of an existing method.
Abstract: A new method for the design of linear-phase robust far-field broadband beamformers using constrained optimization is proposed. In the method, the maximum passband ripple and minimum stopband attenuation are ensured to be within prescribed levels, while at the same time maintaining a good linear-phase characteristic at a prescribed group delay in the passband. Since the beamformer is intended primarily for small-sized microphone arrays where the microphone spacing is small relative to the wavelength at low frequencies, the beamformer can become highly sensitive to spatial white noise and array imperfections if a direct minimization of the error is performed. Therefore, to limit the sensitivity of the beamformer the optimization is carried out by constraining a sensitivity parameter, namely, the white noise gain (WNG) to be above prescribed levels across the frequency band. Two novel design variants have been developed. The first variant is formulated as a convex optimization problem where the maximum error in the passband is minimized, while the second variant is formulated as an iterative optimization problem and has the advantage of significantly improving the linear-phase characteristics of the beamformer under any prescribed group delay or linear-array configuration. In the second variant, the passband group-delay deviation is minimized while ensuring that the maximum passband ripple and stopband attenuation are within prescribed levels. To reduce the computational effort in carrying out the optimization, a nonuniform variable sampling approach over the frequency and angular dimensions is used to compute the required parameters. Experiment results show that beamformers designed using the proposed methods have much smaller passband group-delay deviation for similar passband ripple and stopband attenuation than a modified version of an existing method.

Journal ArticleDOI
TL;DR: A convergence comparison of several nonlinear approaches for the phase retrieval problem involving regularizations with sparsity constraints finds the best simulation results are obtained by the first method for the various noise levels and initializations investigated.
Abstract: The phase retrieval process is a nonlinear ill-posed problem. The Fresnel diffraction patterns obtained with hard x-ray synchrotron beam can be used to retrieve the phase contrast. In this work, we present a convergence comparison of several nonlinear approaches for the phase retrieval problem involving regularizations with sparsity constraints. The phase solution is assumed to have a sparse representation with respect to an orthonormal wavelets basis. One approach uses alternatively a solution of the nonlinear problem based on the Frechet derivative and a solution of the linear problem in wavelet coordinates with an iterative thresholding. A second method is the one proposed by Ramlau and Teschke which generalizes to a nonlinear problem the classical thresholding algorithm. The algorithms were tested on a 3D Shepp–Logan phantom corrupted by white Gaussian noise. The best simulation results are obtained by the first method for the various noise levels and initializations investigated. The reconstruction errors are significantly decreased with respect to the ones given by the classical linear phase retrieval approaches.

Proceedings ArticleDOI
Akihiko Sugiyama1, Ryoji Miyahara
01 Oct 2013
TL;DR: Comparison of enhanced signal spectrogram with that of clean speech demonstrates superior enhanced signal quality.
Abstract: This paper proposes tapping noise suppression with a new phase-based detection. Phase slope of the input noisy signal is compared with an ideal phase slope obtained from an average of intra-frame slopes along the frequency axis. In order to cope with heavily low-pass characteristics of tapping noise spectrum, phase values are weighted with the magnitude at each frequency point. Phase unwrapping problem is alleviated by use of a rotation vector of frequency domain components. Comparison of enhanced signal spectrogram with that of clean speech demonstrates superior enhanced signal quality.