Implementation of nonlinear FM pulse compression filters using surface wave delay lines

The synthesis of a surface wave pulse compression filter with a non-linear FM dispersion relation to reduce time sidelobes is described. The significance of this synthesis is discussed and a comparison of the use of nonlinear FM and amplitude weighting of linear FM is made with regard to surface wave synthesis.     

Estimation of modulation based on FM-to-AM transduction:two-sinusoid case

A method is described for estimating the amplitude modulation (AM) and the frequency modulation (FM) of the components of a signal that consists of two AM-FM sinusoids. The approach is based on the transduction of FM to AM that occurs whenever a signal of varying frequency passes through a filter with a nonflat frequency response. The objective is to separate the AM and FM of the sinusoids from the amplitude envelopes of the output of two transduction filters, where the AM and FM are nonlinearly combined in the amplitude envelopes. The current scheme is first refined for AM-FM estimation of a single AM-FM sinusoid by iteratively inverting the AM and FM estimates to reduce error introduced in transduction. The transduction filter pair is designed relying on both a time- and frequency-domain characterization of transduction error. The approach is then extended to the case of two AM-FM sinusoids by essentially reducing the problem to two single-component AM-FM estimation problems. By exploiting the beating in the amplitude envelope of each filter output due to the two-sinusoidal input, a closed-form solution is obtained. This solution is also improved upon by iterative refinement. The AM-FM estimation methods are evaluated through an error analysis and are illustrated for a wide range of AM-FM signals     

Automatic Cancellation Method of Multipath Echo Distortion in FM Broadcasting Receivers

An automatic canceling method of multipath echo distortion in FM broadcasting receiver is proposed. The cancelling system is composed of two additional parts to the conventional receiver: a programable transversal filter operating at linear IF stage and a microcomputer. The microcomputer calculates echo parameters (relative echo amplitudes, delay times and phase differences between direct wave and echoes at the carrier frequency), from the amplitude-frequency characteristics between sending station and IF stage of the receiver, which is obtained from both outputs of an IF signal envelope detector and FM demodulator. The microcomputer sends control signals to the weighting circuits of the transversal filter. The output signal of the transversal filter is fed back and added to the incoming signal. Tap weighting adjustment is continued until a flat amplitude-frequency characteristics is obtained. The results of computer simulation show this method works well for various echo conditions and the distortion can be completely eliminated.     

A weighted least squares algorithm for quasi-equiripple FIR and IIRdigital filter design

It has been demonstrated by several authors that if a suitable frequency response weighting function is used in the design of a finite impulse response (FIR) filter, the weighted least squares solution is equiripple. The crux of the problem lies in the determination of the necessary least squares frequency response weighting function. A novel iterative algorithm for deriving the least squares frequency response weighting function which will produce a quasi-equiripple design is presented. The algorithm converges very rapidly. It typically produces a design which is only about 1 dB away from the minimax optimum solution in two iterations and converges to within 0.1 dB in six iterations. Convergence speed is independent of the order of the filter. It can be used to design filters with arbitrarily prescribed phase and amplitude response     

Blind Hopping Phase Estimator in Frequency‐Hopped FM and BFSK Systems

A blind hopping phase estimator is proposed for the demodulation of received signals in frequency‐hopping spread spectrum systems. The received signals are assumed to be bandwidth limited with a shaping filter, modulated as frequency modulation (FM) or binary frequency shift keying (BFSK), and hopped by predetermined random frequency sequences. In the demodulation procedure in this paper, the hopping frequency tracking is accomplished by choosing a frequency component with maximum amplitude after taking a discrete Fourier transform, and the hopping phase estimator performs the conjugated product of two consecutive signals and moving‐average filtering. The probability density function and Cramer‐Rao low bound (CRLB) of the proposed estimator are evaluated. The proposed scheme not only is very simple to implement but also performs close to the CRLB in demodulating hopped FM/BFSK signals.     

CDMA信号时差频差联合估计算法研究

针对CDMA信号时域频域同时重叠的特征,分析了不同用户的时差分布对互模糊函数计算的影响,总结了直接互模糊相关法的不足之处。在此基础上,提出了一种解扩后互模糊相关算法,并将其与直接互模糊相关算法进行性能比较,推导了解扩后互模糊相关对信噪比的改善值。仿真结果验证了解扩后互模糊相关算法具有抑制其他用户干扰和改善输出信噪比的优点。     

SAW chirp filter performance above 1 GHz

The performance of a practical compact SAW dispersive filter with a bandwidth in excess of 500 MHz and centered at 1.3 GHz is described. The linear FM chirp filter was fabricated and serially reproduced by electron-beam microfabrication. The effect of an externally implemented weighting function on the time domain performance is discussed. The filter has been found suitable for pulse compression in a high-resolution monopulse radar. Replication of filter sets with identical characteristics is discussed.     

T2 restoration and noise suppression of hybrid MR images using Wiener and linear prediction techniques

The authors address the problem of enhancing hybrid magnetic resonance (MR) images degraded by T2 effects and additive measurement noise. To reduce imaging time, MR signals are acquired using hybrid imaging (HI) sequences such as rapid acquisition relaxation-enhanced (RARE) and fast spin-echo (FSE). With these techniques, T2 effects act as a distortion filter. This T2 filter affects the signal and results in image spatial resolution and/or contrast loss. Furthermore, the amplitude and phase discontinuities in the T2 filter frequency response function may generate serious ringing artifacts. These distortions will damage image quality and affect object detectability. The authors use the Wiener filter and linear prediction (LP) technique to process HI MR signals in the spatial frequency domain (K-space) and the hybrid domain, respectively. Based on the average amplitude symmetry constraint of the spin echo signal, the amplitude frequency response function of the T2 distortion filter can be estimated and used in the Wiener filter for a global T2 amplitude restoration. Then, the linear prediction technique is utilized to obtain the local signal amplitude and phase estimates around the discontinuities of the frequency response function of the T2 filter. These estimates are used to make local amplitude and phase corrections. The effectiveness of this combined technique in correcting T2 distortion and reducing the measurement noise is analyzed and demonstrated using experiments on both phantoms and human studies.     

The resolution of closely spaced signals

This paper presents a method for resolving signals closely spaced in parameter space in the sense that the parameters of the signals being measured (i.e., time of arrival, frequency, etc.) are close together. A maximum-likelihood method is used to resolveMsignals inK-dimensional space, whereMmay be unknown. The resulting procedure first generates aK-dimensional cross-ambiguity function and then passes this function through aK-dimensional linear filter. The procedure effectively reduces the problem from its original form of optimally searching for a maximum in the(M times K)-dimensional space to searching forMmaxima in theK-dimensional parameter space. The method is obviously sub-optimal; its advantage lies in the relatively simple form of the detection scheme.     

Adaptive Median Filtering for Impulse Noise Elimination in Real-Time TV Signals

A two-dimensional (3 × 3) median filter with controlled turn-on is employed in an adaptive fashion for selective removal of impulse noise interference in real-time television signals. Various filters and threshold conditions are selected from programmed PROM's as a function of impulse noise detected and counted during a vertical interval window period. The adaptive median filter is used in conjunction with a satellite FM communication downlink system and can effectively improve the onset of threshold performance by about 3 dB carrier-to-noise ratio. The particular FM system discussed employs a multiplexed analog component, MAC, formatted TV signal. Use of the filter for standard NTSC composite signals is indicated.     

Digital signal processing for video applications

Digital video signal processing is one result of the fast progress in NMOS-VLSI techniques. The attractions of using digital data processing methods in an analog application field are the availability of CAD tools for the design of digital ICs and the integration of digital filter functions. Besides the key components such as microcomputers, A/D, and D/A converters, the digital filter techniques are the most important functions in this application field. It is demonstrated that digital signal processing is not only restricted to amplitude modulated video signals, but also that frequency modulated signals can be processed and methods for FM modulation and demodulation have been developed.     

A fast algorithm for FIR digital filtering with a sum-of-triangles weighting function

A doubly recursive algorithm for time domain convolution with a piecewise linear weighting function is presented that combines the speed of a recursive (IIR) digital filter with the flexibility and ease of design of a nonrecursive (FIR) digital filter. The approach approximates the desired FIR weighting function by a sum-of-triangles weighting function. ForL triangles (or triangle pairs for a linear phase filter) the algorithm is of orderLN. The approximation improves with the number of triangles. A significant advantage of the algorithm compared to FFT filtering or direct convolution is that there is no necessity of a tradeoff between frequency response accuracy and computation time per output point as the data spacing decreases in the filtered signal. The computational complexity is dependent on the number of triangles chosen, not the width of the weighting function, so the algorithm is especially effective for filters with an inherently wide FIR weighting function.     

Weighted least squares near-equiripple approximation of variable fractional delay FIR filters

The weighted least squares (WLS) method is a well-known method for designing a finite impulse response (FIR) filter. And some authors have reported that if a suitable frequency response weighting function is used to design the filter, the WLS method can produce an equiripple result. However, the weighting function for minimax optimality of WLS design is hard to derive analytically. By an iterative method with an adjustable elaborately constructed weighting function, this idea is extended to design a near-equiripple variable fractional delay FIR filter. The proposed method is superior to the fixed-weighting WLS design in the peak absolute error by about 6.6874 dB. The algorithm converges very rapidly. From the simulation, it typically produces a design which is only about 1 dB away from the truly equiripple solution in two iterations and converges to within 0.0056 dB in eight iterations.     

RLS-type two-dimensional adaptive filter with a t-distribution assumption

In this paper we propose a novel two-dimensional adaptive filter based on assumption that the error signal has a t-distribution probability distribution with α degree of freedom to reduce the effect of large amplitude errors. We apply a large weighting function for error signals which have small amplitude and small weighting function for errors, which have large amplitude. By doing so, the effect of the large amplitude error signal to the obtained adaptive parameter can be suppressed. The parameter of the adaptive system is solved using an RLS-like adaptive algorithm. The proposed adaptive algorithm was applied to enhance images that are contaminated with additive noise. The ability of small α t-distribution assumption to reduce various additive noise is higher than that by using large α. The obtained noise reduction (NR) by utilizing small α is even higher than that obtained by applying the recently proposed LMS L-filter. Furthermore, the proposed adaptive algorithm has been applied for image linear prediction. By applying small α, we can obtain lower entropy than that by utilizing large α.     

Statistical AM-FM models, extended Kalman filter demodulation,Cramer-Rao bounds, and speech analysis

A stochastic dynamical system model for describing time signals that are jointly amplitude (AM) and frequency (FM) modulated is presented. The signal is assumed to be bandpass, perhaps originating from a filter bank applied to a broadband signal, and includes the constraint that the magnitude of the complex baseband signal is positive. Motivated by speech processing and the desire for narrowband modulating signals, time is divided into frames, and the modulating signals are smoothly interpolated across each frame. The model allows a detailed characterization of the bandwidth of the modulating signals and the statistical character of the measurement noise. An adaptive estimation algorithm based on extended Kalman filtering ideas for extracting the modulating signals from the measured signal is described and demonstrated on both voiced and unvoiced speech signals. The Cramer-Rao bound on the performance of any estimator is computed     

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.     

Energy separation in signal modulations with application to speechanalysis

An efficient solution to the fundamental problem of estimating the time-varying amplitude envelope and instantaneous frequency of a real-valued signal that has both an AM and FM structure is provided. Nonlinear combinations of instantaneous signal outputs from the energy operator are used to separate its output energy product into its AM and FM components. The theoretical analysis is done first for continuous-time signals. Then several efficient algorithms are developed and compared for estimating the amplitude envelope and instantaneous frequency of discrete-time AM-FM signals. These energy separation algorithms are used to search for modulations in speech resonances, which are modeled using AM-FM signals to account for time-varying amplitude envelopes and instantaneous frequencies. The experimental results provide evidence that bandpass-filtered speech signals around speech formants contain amplitude and frequency modulations within a pitch period     

Error floors of digital FM in simulcast and Rayleigh fading

A digital FM system in the presence of simulcast interference and/or Rayleigh fading exhibits performance floors, where increased signal strength does not reduce the error rate. Results obtained both in the lab, using a radio channel simulator, and in the field show this performance. Observation of signal waveforms indicates that the bit-error rate (BER) floor is a result of FM clicks during deep fades. We present a mathematical model that can be used to calculate the BER floors. The model is based on the calculation of the click shape as a function of amplitude imbalance, frequency offset, and postdetection filter impulse response. The Rayleigh fading is approximated with a two-signal simulcast by substituting the RMS Doppler spread for the simulcast frequency offset. A good match between calculated and measured results is obtained. It is found that the symbol errors caused by simulcast and Rayleigh fading can be treated as independent, and their effect is cumulative     

Modulation Transfer from TDMA On-Off Bursting to FM Carriers in Memoryless Nonlinear Devices, Part II-Baseband Performance

When TDMA and FM carriers are commonly amplified by a memoryless device with nonlinear amplitude and phase characteristics, modulation transfer noise induced by the on-off bursting of the TDMA signals will be present at the FM carriers. In Part II of this paper, closed form expressions of the power spectra of the modulation transfer noise are derived for many special carrier configurations of interest. Numerical examples are also provided in terms of the signal-to-modulation transfer noise ratio, SNR, in the baseband of the FM carriers. It is shown how SNR varies as a function of TDMA burst levels, burst patterns, burst lengths, FM carrier levels, carrier sizes, and device nonlinearities and operating points. With these detailed analyses and numerical examples, insight can be gained into the mechanism of modulation transfer from TDMA on-off bursting to the FM signals. Results presented in this part of the paper should be useful for transmission systems planning and evaluation, and for TDMA burst scheduling.     

Compensation technique for the design of wideband surface-acoustic-wave filters using multistrip couplers

The letter describes a technique which can be used to compensate for losses in amplitude response induced by beam profile reshaping by a multistrip coupler when used with weighted surface-acoustic-wave transducers having large time-band-width products. This is achieved by analytically modifying the conventional weighting of the transducers. Experimental results from two filter designs employing this technique are presented.