This paper presents an ultra-wideband (UWB) bandpass filter using a combination of broadside-coupled structure and lumped-capacitor-loaded shunt stub resonator. The broadside-coupled microstrip-to-coplanar waveguide structure provides an ultra-wide bandpass filtering operation and keeps a good stopband at lower frequencies from DC at the same time. The lumped-capacitor-loaded shunt stub resonator creates two transmission zeros (attenuation poles which can be located at the outsides of the two bandedges of the UWB bandpass filter to improve the out-band performance by selecting a suitable combination of the length of the shunt stubs and the capacitance of the loaded chip capacitors. The filter was designed based on electromagnetic simulation for broadside-coupled structure, microwave circuit simulation and experiments for determining the transmission zeros. The filter was fabricated on a one-layer dielectric substrate. The measured results demonstrated that the developed UWB bandpass filter has good performance: low insertion loss about 0.46 dB and low group delay about 0.26 ns at the center of the passband and very flat over the whole passband, and less than -10 dB reflection over the passband. The implemented transmission zeros, particularly at the low frequency end, dramatically improved the out-band performance, leading the filter satisfy the FCC's spectrum mask not only for indoor but also for outdoor applications. These poles improved also the skirt performance at both bandedges of the filter. A lowpass filter has been also introduced and integrated with the proposed bandpass filter to have a further improvement of the out-band performance at the high frequency end. The filters integrated with lowpass section exhibit excellent filter performance: almost satisfying the FCC's spectrum mask from DC to 18 GHz. The developed UWB bandpass filter has a compact size of 4 cm1.5 cm, or 4.8 cm1.5 cm with lowpass section implemented.

This paper presents a partially resonant active filter based on a digital PWM control circuit with a DSP that can improve the power factor and input current harmonic distortion factor of distributed power supply systems in communications buildings. The steady-state and dynamic characteristics of this active filter are analyzed experimentally and the relationship between the control variables of digital control circuit with the DSP and performance characteristics such as regulation of the output voltage, input power factor, input current harmonic distortion factor, boundaries of stabilities and transient response are defined. Using the partially resonant circuit, the efficiency is over 91%, which is 0.9 point higher than that of non-resonant circuit and the high frequency switching noise is suppressed. Furthermore, the digital control strategy with the DSP proposed in this paper can realize the superior transient response of input current and output voltage for the step change of load, the power factor over 0.99 and total harmonic distortion factor less than 1.1%.

The article describes recent adaptive estimation algorithms over distributed networks. The algorithms rely on local collaborations and exploit the space-time structure of the data. Each node is allowed to communicate with its neighbors in order to exploit the spatial dimension, while it also evolves locally to account for the time dimension. Algorithms of the least-mean-squares and least-squares types are described. Both incremental and diffusion strategies are considered.

Orthogonal frequency division multiplexing (OFDM) is an attractive technique to accomplish wired or wireless broadband communications. Since it has been adopted as the terrestrial digital-video-broadcasting standard in Europe, it has also subsequently been embedded into many broadband communication standards. Many techniques for frame timing and frequency synchronization of OFDM systems have been studied as a result of its increasing importance. We propose a new technique of simultaneously synchronizing frame timing and frequency utilizing matched filters. First, a new short preamble consisting of short sequences multiplied by a DBPSK coded sequence is proposed. Second, we show that the new short preamble results in a new structure for matched filters consisting of a first matched filter, a DBPSK decoder, and a second matched filter. We can avoid the adverse effects of carrier frequency offset (CFO) when frame timing is synchronized because a DBPSK decoder has been deployed between the first and second matched filters. In addition, we show that the CFO can be directly estimated from the peak value of matched filter output. Finally, our simulation results demonstrate that the proposed scheme outperforms the conventional schemes.

This paper proposes a new adaptive filter algorithm for system identification by using an independent component analysis (ICA) technique, which separates the signal from noisy observation under the assumption that the signal and noise are independent. We first introduce an augmented state-space expression of the observed signal, representing the problem in terms of ICA. By using a nonparametric Parzen window density estimator and the stochastic information gradient, we derive an adaptive algorithm to separate the noise from the signal. The proposed ICA-based algorithm does not suppress the noise in the least mean square sense but to maximize the independence between the signal part and the noise. The computational complexity of the proposed algorithm is compared with that of the standard NLMS algorithm. The stationary point of the proposed algorithm is analyzed by using an averaging method. We can directly use the new ICA-based algorithm in an acoustic echo canceller without double-talk detector. Some simulation results are carried out to show the superiority of our ICA method to the conventional NLMS algorithm.

This study employs secret codes and secret keys based on the elliptic curve to construct an elliptic curve cryptosystem with a dynamic access control system. Consequently, the storage space needed for the secret key generated by an elliptic curve dynamic access control system is smaller than that needed for the secret key generated by exponential operation built on the secure filter (SF) dynamic access control system. Using the elliptic curve to encrypt/decrypt on the secure filter improves the efficiency and security of using exponential operation on the secure filter in the dynamic access control system. With the proposed dynamic elliptic curve access control system, the trusted central authority (CA) can add/delete classes and relationships and change the secret keys at any time to achieve an efficient control and management. Furthermore, different possible attacks are used to analyze the security risks. Since attackers can only obtain the general equations for the elliptic curve dynamic access control system, they are unable to effectively perform an elliptic curve polynomial (ECP) conversion, or to solve the elliptic curve discrete logarithm problem (ECDLP). Thus, the proposed elliptic curve dynamic access control system is secure.

This paper proposes a novel robust fundamental frequency (F0) estimation algorithm based on complex-valued speech analysis for an analytic speech signal. Since analytic signal provides spectra only over positive frequencies, spectra can be accurately estimated in low frequencies. Consequently, it is considered that F0 estimation using the residual signal extracted by complex-valued speech analysis can perform better for F0 estimation than that for the residual signal extracted by conventional real-valued LPC analysis. In this paper, the autocorrelation function weighted by AMDF is adopted for the F0 estimation criterion and four signals; speech signal, analytic speech signal, LPC residual and complex LPC residual, are evaluated for the F0 estimation. Speech signals used in the experiments were an IRS filtered speech corrupted by adding white Gaussian noise or Pink noise whose noise levels are 10, 5, 0, -5 [dB]. The experimental results demonstrate that the proposed algorithm based on complex LPC residual can perform better than other methods in noisy environment.

An explicit expression for the impulse response coefficients of the predictive FIR digital filters is derived. The formula specifies a four-parameter family of smoothing FIR digital filters containing the Savitsky-Goaly filters, the Heinonen-Neuvo polynomial predictors, and the smoothing differentiators of arbitrary integer orders. The Hahn polynomials, which are orthogonal with respect to a discrete variable, are the main tool employed in the derivation of the formula. A recursive formula for the computation of the transfer function of the filters, which is the z-transform of a terminated sequence of polynomial ordinates, is also introduced. The formula can be used to design structures with low computational complexity for filters of any order.

This paper discusses a new structure of M-channel IIR perfect reconstruction filter banks. A novel building block defined as a cascade connection of some IIR building blocks and FIR building blocks is presented. An IIR building block is written by state space representation, where we easily obtain a stable filter bank by setting eigenvalues of the state transition matrix into the unit circle. Due to cascade connection of building blocks, we are able to design a system with a larger number of free parameters while keeping the stability. We introduce the condition which obtains the new building block without increasing of the filter order in spite of cascade connection. Additionally, by showing the simulation results, we show that this implementation has a better stopband attenuation than conventional methods.

This paper presents a novel stereophonic acoustic echo canceling scheme without preprocessing. To accurately estimate echo path keeping the high level of performance in echo erasing, this scheme uses two filters, of which one filter is utilized as a guideline which does not erases echo but helps updating of the other filter, which actually erases echo. In addition, we propose a new filter dividing technique to apply to the filter divide scheme, and utilize this as the guideline. Numerical examples demonstrate that the proposed scheme improves the convergence behavior compared to conventional methods both in system mismatch (i.e., normalized coefficients error) and Echo Return Loss Enhancement (ERLE).

In this paper, a new subspace-based speech enhancement algorithm is presented. First, we construct a perceptual filterbank from psycho-acoustic model and incorporate it in the subspace-based enhancement approach. This filterbank is created through a five-level wavelet packet decomposition. The masking properties of the human auditory system are then derived based on the perceptual filterbank. Finally, the prior SNR and the masking threshold of each critical band are taken to decide the attenuation factor of the optimal linear estimator. Five different types of in-car noises in TAICAR database were used in our evaluation. The experimental results demonstrated that our approach outperformed conventional subspace and spectral subtraction methods.

This paper presents the implementation of a 31-tap FIR Hilbert transform digital filter chip used in the digital-IF receivers, to confirm the effectiveness of our new design method. Our design method that we previously reported is based on a computation sharing multiplier using a new horizontal and vertical common subexpression techniques. A 31-tap FIR Hilbert transform digital filter was implemented and fabricated in 0.35 µm CMOS standard cell library. The chip's core contains approximately 33k transistors and occupies 0.86 mm2. The chip also has an operating speed of 70 MHz over. The implementation results show that the proposed Hilbert transformer has a smallest cost factor and so that is a high performance filter.

The objective of this paper is to develop the theoretical foundation to the pilot-assisted channel estimation using delay-time domain windowing for the coherent detection of OFDM signals. The pilot-assisted channel estimation using delay-time domain windowing is jointly used with polynomial interpolation, decision feedback and Wiener filter. A closed-form BER expression is derived. The impacts of the delay-time domain window width, multipath channel decay factor, the maximum Doppler frequency are discussed. The theoretical analysis is confirmed by computer simulation.

This paper proposes a co-channel interference (CCI) suppression scheme employing a frequency-domain nulling filter and turbo equalizer for single-carrier uplink time division multiple access (TDMA) systems. In the proposed scheme, after the received signal is transformed into a frequency-domain signal via fast Fourier transform (FFT), CCI from an adjacent cell is suppressed by the nulling filter. Moreover, the proposed scheme employs a soft canceller and minimum mean square error (SC/MMSE) based turbo equalizer to suppress the performance degradation due to inter-symbol interference (ISI) caused by the nulling filter as well as the ISI induced by fading channel. Computer simulation confirms that the proposed scheme is effective in suppression of CCI compared to the conventional linear frequency-domain equalizer.

A design method for an ultra-wideband bandpass filter (BPF) with four coupled lines has been developed. For demonstration purposes, 50 Ω-matched self-complementary antennas integrated with the ultra-wideband, differential-mode BPF with four coupled lines, a notch filter, and a low-pass filter (LPF) were prepared and tested. An optimized structure for a single-stage, broadside-coupled and edge-coupled four-lines BPF was shown to exhibit up to 170% fractional bandwidth and an impedance transformation ratio of 1.2 with little bandwidth reduction, both analytically and experimentally. Using the optimized structure, 6-stage BPFs were designed to transform the self-complementary antenna's constant input impedance (60πεe- 1/2(Ω)) to 50 Ω without degrading bandwidth. In addition, two types of filter variations--a LPF-embedded BPF and a notch filter-embedded BPF--were designed and fabricated. The measured insertion loss of both filter systems was less than 2.6 dB over the ultra-wideband (UWB) band from 3.1 GHz to 10.6 GHz. The filter systems were embedded in the wideband self-complementary antennas to reject unnecessary radiation over the next pass band and 5-GHz wireless LAN band.

In this paper, we present a directional interpolation filter in which the minimum and maximum pixels in the given window are excluded. Image pixels within a predefined window are ranked and classified as minimum-maximum or exclusive level, and then passed through the interpolation and identity filters, respectively. Extensive simulations show that the proposed filter performs better than other nonlinear filters in preserving desired image features while reducing impulse noise effectively.

It is an innovative idea for modern PLL generation to control the bandwidth proportionally to the reference frequency. Recently, a frequency of the operating clock in microprocessors has been required to be changed frequently and widely in order to manage power consumption and throughput. A new compact switched capacitor (SC) filter which has fully flat response has been developed for adaptive biased PLLs. We have also developed a new digital control method for achieving the wider frequency range. The measured performances of the test chip were good enough for the use in the microprocessors.

A fast fc automatic tuning circuit suitable for WCDMA systems is proposed. The circuit employs master-slave architecture using digitally controlled Gm-C filter for avoiding long transient response. The tuning feedback loop contains a 2-bit up-down counter ADC for fast tuning operation. Furthermore, to avoid degradation of fc tuning accuracy due to reference feedthrough, an analog loop filter with notch located near reference frequency is used. The fast fc automatic tuning circuit is fabricated in a SiGe BiCMOS process. The tuning time within 200 µs is achieved for 35 chips from 2 lots and the standard deviation of 25.5 kHz is obtained for the average fc of 2.12 MHz.

This paper proposes a Mel-Wiener filter to enhance Mel-LPC spectra in the presence of additive noise. The transfer function of the proposed filter is defined by using a first-order all-pass filter instead of unit delay. The filter coefficients are estimated based on minimization of the sum of the square error on the linear frequency scale without applying the bilinear transformation and efficiently implemented in the autocorrelation domain. The proposed filter does not require any time-frequency conversion, which saves a large amount of computational load. The performance of the proposed system is comparable to that of ETSI AFE. The optimum filter order is found to be 3, and thus filtering is computationally inexpensive. The computational cost of the proposed system except VAD is 53% of ETSI AFE.

A 1-V operational sigma-delta modulator with a second-order passive switched capacitor filter is designed and fabricated by using a 90 nm CMOS process. No gate-voltage bootstrapped scheme is adopted to drive analog switches, and the voltage gain of a comparator is chosen to be 94 dB. The experimental results show that the peak SNR reached 68.9 dB with a frequency bandwidth of 40 kHz when the clock was 40 MHz.