Measurement of time delays in multipath systems

In a recent letter, Gouriet proposed a method of measuring time delay, which uses the fact that the time delay of a linear system is the group delay at zero frequency. Based on the same method, a system is proposed in this letter to measure the time delays in a linear system which provides more than one transmission path between a single output and a single input. The system assumes that a rough knowledge of the magnitudes of time delays is available and that the individual multipath responses do not overlap in the time domain.     

Output effect of a quadrature receiver with an input band-pass filter at reception of a phase-keyed wideband signal

An output effect of a quadrature correlation receiver with an input band-pass filter at reception of a phase-keyed wideband signal is considered. Analytical expressions for output effect of a receiver proportional to the absolute value of a complex envelope of a cross-correlation function for received and reference signals are obtained. Dependences of cross-correlation function absolute value on a time delay, frequency mismatch of a received and reference signals, and passband width of input filter are analyzed. It is shown, that cross-correlation function’s maximum is reached in case of non-zero time delay of a received signal when there is no frequency mismatch between input and reference signals. The time delay value corresponding to cross-correlation function’s maximum depends on passband width of input filter.     

Algorithms for a Sparse Reconfigurable Adaptive Filter and a Photonic Switch Architecture

A nonblocking photonic switch can be used to implement a tapped delay line with a large number of adaptive weights and a wide range of time delays. An advantage of using optical tapped delay lines for adaptive filtering is that the operating frequency can be quite high, in the 10?100 GHz range. We present a sparse reconfigurable adaptive filter (SRAF) based on a photonic switch with an input/output connection architecture that can be represented by a matrix of adaptive weights. This unique parallel structure can be reconfigured in an adaptive manner to implement a sparse filter impulse response for use in many applications. We consider an adaptive algorithm for this filter that chooses the input and output delays using a cross-correlation-based approach and connects these delays by weights that are adapted using a gradient algorithm. An alternative adaptive algorithm is also considered that is based on a system identification formulation where the weights are first adapted, and then the appropriate delay combinations are chosen. A search algorithm for implementing the connection constraint required by the SRAF is also discussed whereby each input is connected to only one output at any moment. Computer simulation examples are presented to illustrate the behavior of the filter for a system identification model.      

A Generic Open-Loop Algorithm for Three-Phase Grid Voltage/Current Synchronization With Particular Reference to Phase, Frequency, and Amplitude Estimation

This paper presents a new open-loop architecture for three-phase grid synchronization based on moving average and predictive filters, where accurate measurements of phase, frequency, and amplitude are carried out in real time. Previous works establish that the fundamental positive sequence vector of a set of utility voltage/current vectors can be decoupled using Park's transformation and low-pass filters. However, the filtering process introduces delays that impair the system performance. More specifically, when the input signal frequency is shifted above the nominal, a nonzero average steady-state phase error appears in the measurements. To overcome such limitations, a suitable combination of predictive and moving average finite impulse response (FIR) filters is used by the authors to achieve a robust synchronization system for all input frequencies. Moving average filters are linear phase FIR filters that have a constant time delay at low frequencies, a characteristic that is exploited to good effect to design a predictive filter that compensates such time delays, enabling zero steady-state phase errors for shifted input frequencies. In summary, the main attributes of the new system are its good frequency adaptation, good filtering/transient response tradeoff, and the fact that its dynamics is independent of the input vector amplitude. Comprehensive experimental results validate the theoretical approach and the high performance of the proposed synchronization algorithm.      

A CMOS continuous-time Gm-C filter for PRML read channelapplications at 150 Mb/s and beyond

Design techniques for equiripple phase CMOS continuous-time filters are presented, and their integration within a partial-response maximum likelihood (PRML) disk drive read channel is discussed. A programmable seven-pole two asymmetric zero filter implementation is described based on a new transconductance (G_m) cell. The impact of integrator finite output impedance, excess phase, and other implementation related nonidealities is discussed. A filter tuning circuit that requires an accurate time base but no external components is presented. The filter has a cutoff frequency (f_c) range of 6-43 MHz, where f_c is the -3 dB point of the magnitude transfer function with the two zeros set to infinity. Also, with finite zeros it is able to provide up to 12 dB of boost which is defined as the maximum value of the magnitude transfer function referred to dc. The group delay ripple stays within ±2% for frequencies below 1.75 f _c. The cutoff frequency exhibits a 650 ppm/°C temperature dependency and a variation of ±1%/V with the power supply. Total harmonic distortion (THD) values are below -40 dB at twice the nominal operating input voltage (V_nominal=320 mV peak-to-peak differential), and the dynamic range exceeds 60 dB (for a maximum input signal of 640 mV peak-to-peak differential, maximum bandwidth setting, and no boost). Both the filter and a tuning circuit were implemented in a 0.6-μm single-poly triple-metal n-well CMOS process. They consume 90 mW from a single 5 V power supply and occupy an area of 0.8 mm²      

Synthesis of a first-order passive complex filter with band-pass/band-elimination characteristics

In this paper, a new passive complex filter is proposed. In the same fashion as the traditional real band-pass filter, the proposed circuit is also designed through the frequency transformation. In order to obtain the prototype complex filter suitable for passive realization, the extended lowpass–highpass transformation or the bilinear lowpass–lowpass transformation different from the well-known frequency shifting method is adopted. These transformations give two filters; however, the final circuits become identical to each other. The proposed circuit has two output terminals. One is the complex band-pass output, and the other is for the complex band-elimination. The proposed circuit includes a capacitor, an autotransformer or an inductor and termination resistors only. Because the proposed circuit has terminating resistors at both of the input and the output sides, it is suitable for high frequency application. As an example, two first-order complex filters which operate in 100 kHz and 10 MHz are designed and their frequency responses are measured. It is shown that the measured frequency responses agree with the theoretical ones through experiment.     

When has a decision-directed equalizer converged?

Adaptive decision-directed equalizers can become locked for long periods onto incorrect equilibria. A test involving data available at the equalizer output is proposed for determining whether an equilibrium is correct or not, up to a fixed overall delay. If an independent sequence of random variables taking values ±1 is the input to a finite impulse response filter, and the output of the filter is passed through a slicer, then the slicer output is uncorrelated if and only if it is a delayed version of the filter input. An analogous result for M-ary rather than binary data is outlined     

A min, + system theory for constrained traffic regulation and dynamic service guarantees

By extending the system theory under the (min, +) algebra to the time-varying setting, we solve the problem of constrained traffic regulation and develop a calculus for dynamic service guarantees. For a constrained traffic-regulation problem with maximum tolerable delay d and maximum buffer size q, the optimal regulator that generates the output traffic conforming to a subadditive envelope f and minimizes the number of discarded packets is a concatenation of the g-clipper with g(t) = min[f(t+ d), f (t)+q] and the maximal f-regulator. The g-clipper is a bufferless device, which optimally drops packets as necessary in order that its output be conformant to an envelope g. The maximal f-regulator is a buffered device that delays packets as necessary in order that its output be conformant to an envelope f. The maximal f-regulator is a linear time-invariant filter with impulse response f, under the (min, +) algebra. To provide dynamic service guarantees in a network, we develop the concept of a dynamic server as a basic network element. Dynamic servers can be joined by concatenation, "filter bank summation," and feedback to form a composite dynamic server. We also show that dynamic service guarantees for multiple input streams sharing a work-conserving link can be achieved by a dynamic service curve earliest deadline scheduling algorithm, if an appropriate admission control is enforced.     

Synthesis of One-Input M-Output Optical FIR Lattice Circuits

This paper presents a one-input M-output (1 x M)configuration and a synthesis algorithm for realizing an optical finite impulse response (FIR) lattice filter having M output channels (M ges 2). The circuit configuration has a multilayer structure consisting of multiple Mach-Zehnder interferometers with delay time differences of zero or Deltatau. It is a natural extension of the conventional two-port optical FIR lattice circuit (M = 2) . The proposed synthesis algorithm is based on factorizations of the paraunitary total transfer matrix. It realizes output responses having highest orders equal to the number of stages for all output channels. It is shown that such a synthesis algorithm maximizes the degrees of freedom for designing output responses. Two kinds of five-channel interleave filters are demonstrated as synthesized examples: one has a flat group delay response while the other has an inclined group delay response.     

乘积型指数滤波器及其在多目标时延估计中的应用

指数滤波器是一类新构造出来的输出信噪比和目标时延分辨力随指数变化的滤波器,该滤波器在损失一定输出信噪比的前提下可以有效提高目标时延分辨力,从而提高目标时延估计精度,但仅采用单个指数滤波器仍存在输出信噪比和目标时延分辨力均达不到实际需求的情况。在乘积型高阶模糊函数乘积运算的启发下,在指数滤波器的基础上提出了一种新的乘积型指数滤波器,并分析了该乘积型指数滤波器的输出信噪比及目标时延分辨力等性能。仿真实验表明,所提的乘积型指数滤波器在低信噪比情况下可以更有效提高多目标时延估计精度,且算法简单易于实现,适用于背景复杂的多目标参数估计任务。     

Parallel-coupled microstrip filters with over-coupled end stages for suppression of spurious responses

In a parallel-coupled microstrip filter, end stages with over-coupling are designed to suppress the unwanted responses at twice the passband frequency (2f/sub 0/). The inherent transmission zero of an over-coupled input/output stage is shown tunable. It is found that increasing the image impedance of the filter sections can further enhance the suppression. The designed bandpass filters thus have a wide upper stopband and improved passband response symmetry. Measured results of fabricated circuits show that the idea works very well.     

Improved Design Equations of the Tapped-Line Structure for Coupled-Line Filters

Coupled-line filters with tapped input and output are studied in this letter. The tapped-line structures are used for transformation of the filter impedances to lower values. This makes the filter more suitable for manufacture. However, the variation of the transformed impedances around the passband central frequency, due to the tapped feeds, distorts the filter responses. In this letter, design equations are proposed to improve the design by taking into account this variation. Finally, a filter design example is given to justify the study     

Scattering function estimation

The estimation of the scattering function of a random, zero-mean, homogeneous, time-variant, linear filter is considered. The sum of the random filter output and independent noise is the input to an estimator. The estimator structure is equivalent to a bank of linear filters followed by squared-envelope detectors; the envelope detector outputs are the input to a final linear filter. The estimator output is shown to be an unconstrained linear operation on the ambiguity function of the estimator input. Except for a bias term due to the additive noise, the mean of the estimator output is an unconstrained linear operation on the scattering function of the random filter. The integral variance of the output is found for a Gaussian channel. The mean and variance clearly indicate the tradeoff between resolution and variance reduction obtained by varying the estimator structure. For any well-behaved channel it is shown that an effectively unbiased estimate of the scattering function can be obtained if the input signal has both sufficient energy and enough time and frequency spread to resolve the random filter; the random filter is not required to be underspread. The variance of an estimate can be further reduced by increasing the time or frequency spread of the transmitted signal.     

A Discrete‐Amplitude Pulse Width Modulation for a High‐Efficiency Linear Power Amplifier

A new discrete‐amplitude pulse width modulation (DAPWM) scheme for a high‐efficiency linear power amplifier is proposed. A radio frequency (RF) input signal is divided into an envelope and a phase modulated carrier. The low‐frequency envelope is modulated so that it can be represented by a pulse whose area is proportional to its amplitude. The modulated pulse has at least two different pulse amplitude levels in order that the duty ratios of the pulse are kept large for small input. Then, an RF pulse train is generated by mixing the modulated envelope with the phase modulated carrier. The RF pulse train is amplified by a switching‐mode power amplifier, and the original RF input signal is restored by a band pass filter. Because duty ratios of the RF pulse train are kept large in spite of a small input envelope, the DAPWM technique can reduce loss from harmonic components. Furthermore, it reduces filtering efforts required to suppress harmonic components. Simulations show that the overall efficiency of the pulsed power amplifier with DAPWM is about 60.3% for a mobile WiMax signal. This is approximately a 73% increase compared to a pulsed power amplifier with PWM.     

Nonlinear Techniques for the Improvement of Signal-to-Noise Ratio

We propose a novel circuit for noise rejection, which is composed mainly of a linear amplifier, a nonlinear amplifier, and a filter. The input signal is applied to a linear amplifier, a bandpass limiter, and an envelope detector followed by a low-pass filter. The output of the lowpass filter is multiplied by the output of the bandpass limiter. The difference between the output of the multiplier and linear amplifier is the output of the proposed circuit. It is then indicated that the proposed circuit rejects noise, of which amplitude is narrow-band relative to the desired signal or noise phase.     

A CMOS high-speed data recovery circuit using the matched delaysampling technique

This paper presents a scheme and circuitry for demultiplexing and synchronizing high-speed serial data using the matched delay sampling technique. By simultaneously propagating data and clock signals through two different delay taps, the sampler achieves a very fine sampling resolution which is determined by the difference between the data and clock delays. This high resolution sampling capability of the matched delay sampler can be used in the oversampling data recovery circuit. A data recovery circuit using the matched delay sampling technique has been designed and fabricated in 1.2-μm CMOS technology. The chip has been tested at 417 Mb/s [2.4 ns nonreturn to zero (NRZ)] input data and demultiplexes serial input data into four 104 Mb/s (9.6 ns NRZ) output streams with 800 mW power consumption at 4 V power supply. While recovering data, the sampling clock running at 1/4 of the data frequency is phase-tracking with the input data based on information extracted from a digital phase control circuit     

Carrier induced transient electric fields in a p-i-n InP-InGaAsmultiple-quantum-well modulator

Low-power time-resolved pump-probe measurements made on a InP-In _0.53Ga_0.47As p-i-n multiple-quantum-well optical modulator are presented. The results show a nonzero signal at negative delays whose magnitude and sign exhibit a strong dependence upon the wavelength and bias. The overall behavior after zero delay, in particular the signal rise and fall times, also depends strongly upon these parameters. A theory, based upon electric field variations due to carrier motion in the intrinsic region of the device, is developed to explain the results. It is found that the signal at negative delay is due to hole space charge trapped in the quantum wells, while the fast effects after zero delay are due to the rapid emission of electrons which cause a temporary reduction in potential across the device. This is followed by diffusive conduction within the contact layers which restores the potential to its former value. The rich variety of signal responses is attributed to transient Stark shifts which vary greatly with well position due to the nonuniformity of the dynamic change in the electric field. Calculated signal responses are in excellent agreement with the experimental results     

Robust frequency-selective filtering using weighted myriad filtersadmitting real-valued weights

Weighted myriad smoothers have been proposed as a class of nonlinear filters for robust non-Gaussian signal processing in impulsive noise environments. However, weighted myriad smoothers are severely limited since their weights are restricted to be non-negative. This constraint makes them unusable in bandpass or highpass filtering applications that require negative filter weights. Further, they are incapable of amplifying selected frequency components of an input signal. In this paper, we generalize the weighted myriad smoother to a richer structure: a weighted myriad filter admitting real-valued weights. This involves assigning a pair of filter weights (one positive and the other negative) to each of the input samples. Equivalently, the filter can be described as a weighted myriad smoother applied to a transformed set of samples that includes the original input samples as well as their negatives. The weighted myriad filter is analogous to a normalized linear FIR filter with real-valued weights whose absolute values sum to unity. By suitably scaling the output of the weighted myriad filter, we extend it to yield the so-called scaled weighted myriad filter, which includes (but is more powerful than) the traditional unconstrained linear FIR filter. Finally we derive stochastic gradient-based nonlinear adaptive algorithms for the optimization of these novel myriad filters under the mean square error criterion     

An asymmetric defected ground structure with elliptical response and its application as a lowpass filter

A new asymmetric defected ground structure (DGS) consisting of two square headed slots connected with a rectangular slot transversely under a microstrip line is proposed. In the frequency characteristics of its unit cell, an attenuation zero is observed close to the attenuation pole. The DGS unit is represented by Cauer's T-network, modeled by a third order elliptical lowpass filter and finally equivalent L-C parameters are extracted. Several studies have been accomplished on the influence of the geometric modification of the filter on its characteristics. Better transition sharpness, lower passband insertion loss and broader stopband are observed, compared to popular dumbbell DGSs due to its additional attenuation zero. The two-cell configuration of this DGS produces a three-pole lowpass filter having wide and deep stopband at low cutoff frequency. This is applied under the input and output feed lines of a parallel-coupled bandpass filter for suppressing harmonic frequencies. A lowpass filter is realized using cascaded investigated DGS and dumbbell DGS under a microstrip line.     

YIG Filters as Envelope Limiters

Envelope limiters are used in such applications as FM demodulation and power leveling. Recently, the envelope-limiting properties of yttrium-iron-garnet (YIG) filters were reported for the special cases of unmodulated and pulsed input signals. Measured data is presented hereon the response of a YIG limiter to AM carriers having modulation index of the order of 50 percent. Sinusoidal, square-wave, and low-pass noise modulating signals were used in the measurements. It was found that a YIG filter will give good envelope Iimiting for modulating frequencies in the submegacycle range. At these low frequencies the carrier and the side frequencies are not limited selectively. At higher modulating frequencies where the limiting is frequency selective, the YIG filter will not remove the variations. In fact, in the particular filter tested, the modulation index was increased, rather than decreased, at modulating frequencies greater than about 750 kc/s. A graph is given showing the measured factor of reduction (or increase) of modulation index, as a function of modulating frequency. The response of the limiter as a function of carrier frequency, modulating frequency, and input power is shown by oscilloscope displays produced by sweeping the carrier frequency or input power. In addition, selected photographs of output envelope waveforms are given.