A simultaneous equations method is one of active noise control algorithms without estimating an error path. This algorithm requires identification of a transfer function from a reference microphone to an error microphone containing the effect of a noise control filter. It is achieved by system identification of an auxiliary filter. However, the introduction of the auxiliary filter requires more number of samples to obtain the noise control filter and brings a requirement of some undesirable assumption in the multiple channel case. In this paper, a new simultaneous equations method without the identification of the auxiliary filter is proposed. By storing a small number of input signals and error signals, we avoid this identification. Therefore, we can reduce the number of samples to obtain the noise control filters and can avoid the undesirable assumption. From simulation examples, it is verified that the merits of the ordinary method is also retained in the proposed method.

In this paper, we treat a design problem for IIR digital filters described by rational transfer function in discrete space. First, we form the filter design problem using the modified least-squares (MLS) criterion and express it as the quadratic form with respect to the numerator and denominator coefficients. Next, we show the relaxation method using the Lagrange multiplier method in order to search for the good solution efficiently. Additionally we can check the filter stability when designing the denominator coefficients. Finally, we show the effectiveness of the proposed method using a numerical example.

Two compact and low loss dual-mode filters are proposed by using degenerate modes of slotted triangular microstrip patch resonators. The geometrical size and radiation loss of the triangular patch are reduced simultaneously by loading both horizontal and vertical slots. The resonant frequencies of two degenerate modes can be easily controlled by varying the dimensions and positions of the slots. A two-pole dual-mode filter operating at 3.94 GHz with a fractional bandwidth of 4.3% is designed, fabricated, and measured. The measured results verify well the theoretical predictions.

We consider a watermark application to assist in the integrity maintenance and verification of the associated images. There is a great benefit in using WM in the context of authentication since it does not require any additional storage space for supplementary metadata, in contrast with cryptographic signatures, for instance. However there is a fundamental problem in the case of exact authentication: How to embed a signature into a cover message in such a way that it would be possible to restore the watermarked cover image into its original state without any error? There are different approaches to solve this problem. We use the watermarking method consisting of modulo addition of a mark and investigate it in detail. Our contribution lies in investigating different modified techniques of both watermark embedding and detection in order to provide the best reliability of watermark authentication. The simulation results for different types of embedders and detectors in combination with the pictures of watermarked images are given.

This paper proposes a parallel coupled microstrip bandpass filter (BPF) with ring Electromagnetic Bandgap (EBG) cells on the middle layer for spurious suppression. The ring EBG cells of the middle layer add a good stopband-rejection mode to the second harmonics of the parallel coupled microstrip BPF with suppression of over -50 dB, without affecting the center frequency and insertion loss of the original designed BPF. The design of ring EBG cells is presented and verified by the experimented results.

This paper presents a design procedure of multilayer single-input/single-output dual-band filters of small size fabricated in high-permittivity LTCC (Low Temperature Co-fired Ceramic) substrates. The present dual-band filter, with pass bands of 950 MHz and 1.9 GHz, consists of two single-band filters each one of which has a matching circuit for controlling the input impedance in the pass band of the counterpart. Discussed first in this paper is a dual-band filter that has the matching circuit connected to the single-band filters externally. Then, a dual-band filter with the matching circuit embedded in its body is investigated together with metallization patterns of the matching elements. Presented finally is a very small single-input/single-output dual-band filter suitable for dual-band wireless communication systems.

In this paper a newly designed internally-coupled asymmetric stepped-impedance resonator (SIR) bandpass filter (BPF) is proposed. The asymmetric SIR structure not only can effectively reduce the circuit size but also can provide two flexibly tunable transmission zeros near the lower and upper passband edges. The first transmission zero is due to the series resonance of the quarter-wavelength open stepped-impedance stub, and the second one is produced by anti-parallel coupling between adjacent SIRs. The proposed BPF was fabricated and simulated using the commercial software HFSS, and agreement between the measured and simulated results was observed. A 0.9-dB insertion loss and a shape factor of 3.6 were achieved in the passband, thus indicating that the proposed filter structure is of practical value.

A novel method is proposed to synthesize dual-band bandpass filters (BPFs) from a prototype lowpass filter. By implementing successive frequency transformations and circuit conversions, a new filter topology is obtained which consists of only admittance inverters and series or shunt resonators, and is thereby easy to be realized by using conventional distributed elements. A microstrip dual-band BPF with central frequencies of 1.8 GHz and 2.4 GHz is designed and fabricated using microstrip lines and stubs. The simulated and measured results show a good agreement and validate thereby the proposed theory.

In ASR (Automatic Speech Recognition) applications, one of the most important issues in the real-time beamforming of microphone arrays is the inability to capture the whole acoustic dynamics via a finite-length of data and a finite number of array elements. For example, the reflected source signal impinging from the side-lobe direction presents a coherent interference, and the non-minimal phase channel dynamics may require an infinite amount of data in order to achieve perfect equalization (or inversion). All these factors appear as uncertainties or un-modeled dynamics in the receiving signals. Traditional adaptive algorithms such as NLMS that do not consider these errors will result in performance deterioration. In this paper, a time domain beamformer using H∞ filtering approach is proposed to adjust the beamforming parameters. Furthermore, this work also proposes a frequency domain approach called SPFDBB (Soft Penalty Frequency Domain Block Beamformer) using H∞ filtering approach that can reduce computational efforts and provide a purified data to the ASR application. Experimental results show that the adaptive H∞ filtering method is robust to the modeling errors and suppresses much more noise interference than that in the NLMS based method. Consequently, the correct rate of ASR is also enhanced.

A novel class of microstrip bandpass filter is configured using the impedance transformers and an improved stepped impedance resonator (SIR). This SIR is composed of a central narrow strip section with an aperture on ground and two wide strip sections at the two sides. This low-high-low SIR resonator has a promising capability in achieving an extremely large ratio of first two resonant frequencies for design of a bandpass filter with ultra-broad stopband. The two quarter-wavelength transformers with low and high impedances, referred as to impedance- and admittance-inverters, are modeled and utilized as alternative types of inductive and capacitive coupling elements with highly tightened degrees for wideband filter design. After extensive investigation is made on the two transformers and the proposed SIR, the two novel bandpass filters are constructed, designed and implemented. Two sets of predicted and measured frequency responses over a wide frequency range both quantitatively exhibit their several attractive features, such as ultra-broad stopband with deep rejection and broadened dominant passband with low insertion loss.

This paper addresses a speech recognition problem in non-stationary noise environments: the estimation of noise sequences. To solve this problem, we present a particle filter-based sequential noise estimation method for front-end processing of speech recognition in noise. In the proposed method, a noise sequence is estimated in three stages: a sequential importance sampling step, a residual resampling step, and finally a Markov chain Monte Carlo step with Metropolis-Hastings sampling. The estimated noise sequence is used in the MMSE-based clean speech estimation. We also introduce Polyak averaging and feedback into a state transition process for particle filtering. In the evaluation results, we observed that the proposed method improves speech recognition accuracy in the results of non-stationary noise environments a noise compensation method with stationary noise assumptions.

A pitch-synchronous (PS) auditory feature extraction method based on ZCPA (Zero-Crossings Peak-Amplitudes) was proposed previously and showed more robustness over a conventional ZCPA and MFCC based features. In this paper, firstly, a non-linear adaptive threshold adjustment procedure is introduced into the PS-ZCPA method to get optimal results in noisy conditions with different signal-to-noise ratio (SNR). Next, auditory masking, a well-known auditory perception, and modulation enhancement that simulates a strong relationship between modulation spectrums and intelligibility of speech are embedded into the PS-ZCPA method. Finally, a Wiener filter based noise reduction procedure is integrated into the method to make it more noise-robust, and the performance is evaluated against ETSI ES202 (WI008), which is a standard front-end for distributed speech recognition. All the experiments were carried out on Aurora-2J database. The experimental results demonstrated improved performance of the PS-ZCPA method by embedding auditory masking into it, and a slightly improved performance by using modulation enhancement. The PS-ZCPA method with Wiener filter based noise reduction also showed better performance than ETSI ES202 (WI008).

In deep sub-micrometer CMOS process, owing to the thin gate oxide and small subthreshold voltage, the leakage current becomes more and more serious. The leakage current has made the impact on phase-locked loops (PLLs). In this paper, the compensation circuits are presented to reduce the leakage current on the charge pump circuit and the MOS capacitor as the loop filter. The proposed circuit has been fabricated in 0.13-µm CMOS process. The power consumption is 3 mW and the die area is 0.270.3 mm2.

In this paper, we present a numerical method for the equiripple approximation of minimum phase FIR digital filters. Many methods have been proposed for the design of such filters. Many of them first design a linear phase filter whose length is twice as long, and then factorize the filter to obtain the minimum phase. Although these methods theoretically guarantee its optimality, it is difficult to control the ratio of ripples between different bands. In the conventional lowpass filter design, for example, when different weights are given for its passband and stopband, one needs to iteratively design the filter by trial and error to achieve the ratio of the weights exactly. To address this problem, we modifies well-known Parks-McClellan algorithm and make it possible to directly control the ripple ratios. The method iteratively solves a set of linear equations with controlling the ratio of ripples. Using this method, the equiripple solutions are obtained quickly.

This paper describes a numerical design procedure of element values of RC polyphase filters with equal minima in stopband and equal ripple in passband. Determination of element values of RC polyphase filters with equal-ripple characteristic have not been solved to the best knowledge of the authors. There found a paper tackling with the problem; however, it can only give sub-optimal solutions via numerical calculation [3]. We propose a numerical element value design procedure for RC polyphase filters with equi-ripple gain in both stopband and passband by using the coefficient matching method. Some design examples are given.

A coplanar type lumped-element 6-pole microwave Chebyshev bandpass filter (BPF) of center frequency (f0) 2.0 GHz and fractional bandwidth (FBW) 1.0 % was designed. For the design method, theory of direct coupled resonator filters using K-inverters was employed. Coplanar type lumped-element BPFs are composed of a meander-line L and interdigital C elements. The frequency response was simulated and analyzed using an electromagnetic field simulator (Sonnet-EM). Further, the changes in f0 and FBW of the BPF were also realized by the mechanical tuning method.

A novel high temperature superconducting interdigital bandpass filter is proposed by using coplanar waveguide quarter-wavelength resonators. The CPW resonators are arranged in parallel, and consequently the filter becomes very compact. The filter is a 5-pole Chebyshev BPF with a midband frequency of 5.0 GHz and an equal-ripple fractional bandwidth of 3.2%. It is fabricated using a YBCO film deposited on an MgO substrate. The measured filtering characteristics agree well with EM simulations and show a low insertion loss in spite of the small size of the filter.

R-regular Mth band filters are an important class of digital filters and are used in constructing Mth-band wavelet filter banks, where the regularity is essential. But this kind of filter has larger stopband peak errors compared with a minimax filter of the same length. In this paper, peak errors in stopband of R-regular 4th-band filters are reduced by means of superimposing two filters with successive regularities. Then the stopband peak errors in the resulting filters are compared with the original ones. The results show that the stopband peak errors are reduced significantly in the synthesized filter that has the same length as the longer one of the two original filters, at the cost of regularity.

A parallel notch filter (PNF) for eliminating a sinusoidal signal whose frequency and phase are unknown, has been proposed previously. The PNF achieves both fast convergence and high estimation accuracy when the step-size for adaptation is appropriately determined. However, there has been no discussion of how to determine the appropriate step-size. In this paper, we derive the convergence condition on the step-size, and propose an adaptive algorithm with variable step-size so that convergence of the PNF is automatically satisfied. Moreover, we present a new filtering structure of the PNF that increases the convergence speed while keeping the estimation accuracy. We also derive a variable step-size scheme for the new PNF to guarantee the convergence. Simulation results show the effectiveness of the proposed method.

Novel microstrip dual-band bandpass filters with controllable fractional bandwidths and good in-between isolation are presented and implemented. A half-wavelength stepped-impedance resonator is firstly characterized, aiming at producing the two resonant frequencies at 2.4 and 5.2 GHz. Two types of coupled microstrip lines in the parallel and anti-parallel formats are then investigated in terms of unified equivalent J-inverter network. Extensive results are derived to quantitatively show their distinctive frequency-distributed coupling performances under different coupling lengths. The coupling degrees of these two coupled lines at the two resonances are properly adjusted to achieve the dual-passband response with varied or tunable bandwidths. In addition, the parallel coupled line is modeled to bring out a transmission zero between the two resonances so as to achieve the good in-between isolation. The three two-stage bandpass filters are initially designed to exhibit their dual-band response with changeable dual-band bandwidths. A three-stage dual-band filter is in final optimally designed and its predicted performance is confirmed in experiment.