In this paper we report a 3rd-order Gm-C filter based on pseudo-differential continuous-time transconductors for applications in low-voltage systems over VHF range. By using a 0.18 µm pure digital CMOS process, a prototype low pass filter with -3 dB frequency programmable from 38 MHz to 213 MHz confirms the feasibility of the proposed filter in applications such as data storage systems.

The actual sound environment system exhibits various types of linear and non-linear characteristics, and it often contains an unknown structure. Furthermore, the observations in the sound environment are often in the level-quantized form. In this paper, a method for estimating the specific signal for stochastic systems with unknown structure and the quantized observation is proposed by introducing a system model of the conditional probability type. The effectiveness of the proposed theoretical method is confirmed by applying it to the actual problem of psychological evaluation for the sound environment.

Jacobian Adaptation (JA) has been successfully used in Automatic Speech Recognition (ASR) systems to adapt the acoustic models from the training to the testing noise conditions. In this work we present an improvement of JA for speaker verification, where a specific training noise reference is estimated for each speaker model. The new proposal, which will be referred to as Model-dependent Noise Reference Jacobian Adaptation (MNRJA), has consistently outperformed JA in our speaker verification experiments.

This paper proposes a new theory and design method for a class of recombination nonuniform filter banks (RNFBs) with linear phase (LP) filters. In a uniform filter bank (FB), consecutive channels are merged by sets of transmultiplexers (TMUXs) to realize a nonuniform FB. RNFBs with LP analysis/synthesis filters are of great interest because the analysis filters for the partially reconstructed signals, through merging, are LP and hence less phase distortions are introduced to the desired signals. We analyze the spectrum supports of the analysis filters of these LP RNFBs. The conditions on the uniform FB and recombination TMUXs of an LP RNFB with good frequency characteristics are determined. These conditions are relatively simple to be satisfied and the uniform FB and recombination TMUXs can be designed separately without much degradation in performance. This allows dynamically recombination of different number of channels in the original uniform FB to give a flexible and time-varying frequency partitioning. Using these results, a method for designing a class of near-perfect-reconstruction (NPR) LP RNFBs with cosine roll-off transition band using the REMEZ algorithm is proposed. A design example is given to show that LP RNFBs with good frequency responses and reasonably low reconstruction errors can be achieved.

In this letter, we propose a 2-D receiver structure for multicarrier code division multiple access (MC-CDMA) systems with the reduced-rank multistage Wiener filter. Due to the fast convergence property of the reduced-rank processing, it outperforms MMSE-based receivers with the classical Wiener solution, which is estimated by using a limited number of samples.

A millimeter wave BPF constructed from the WG mode dielectric disk resonators is presented. The design chart for the high Q WG mode resonator is obtained from Qu calculation of some WG modes. By using the design chart, high Q WG mode resonator having no influence of unwanted higher order resonances is designed. Designed resonators have different diameter and various Resonance Frequency Separation respectively. A 3 stage maximally flat BPF is constructed so that each resonator may be coupled laterally on the edge of the disk. Designed center frequency is 62.47 GHz and 3 dB bandwidth is 100 MHz. As a result, this BPF has insertion loss of 1.5 dB and some spurious responses which were existed conventional WG mode BPF are reduced considerably.

This paper presents a new multistage comb-rotated sinc (RS) decimator with a sharpened magnitude response. Novelty of this paper is that the multistage structure has more design parameters that provides additional flexibility to the design procedure. It uses different sharpening polynomials and different cascaded comb filters at different stages. As the comb filters at the latter stages are of lower order than that of the original comb filter, the use of more complex sharpening polynomials at latter stages is possible. This leads to an improvement of the frequency characteristic without a significant increase in the complexity of the overall filter. The comb filter of the first stage is realized in a non-recursive form and can be implemented in a computationally efficient form by making use of the polyphase decomposition of the transfer function in which the subfilters operate at a lower rate that depends on the down-sampling factor employed in the first stage. In addition, both multipliers of the rotated sinc (RS) filter of the second stage work at a lower rate.

This paper presents a highly parallel architecture for deblocking filter in H.264/AVC. We adopt various parallel schemes in memory sub-system and datapath. A 2-dimensional parallel memory scheme is employed to support efficient parallel access in both horizontal and vertical directions in order to speed up the whole filtering process. This parallel memory also eliminates the need for a transpose circuit. In the datapath, an algorithm optimization is performed to implement parallel filtering with hardware reuse. Pipeline techniques are also adopted to improve the throughput of filtering operations. Our design is implemented under TSMC 0.18 µm technology. Results show that the core size is 0.821.13 mm2 when the maximum frequency is 230 MHz. Compared to other existing architectures, our design has advantages in both speed and area.

Two types of miniaturized high-temperature superconducting filters are described in this paper. The first type is developed by using small-sized microstrip spiral resonators, and the second type by coplanar waveguide quarter-wavelength resonators. The filters have significantly reduced size compared with many previous HTS filters. They are designed by employing an electromagnetic simulator in combination with appropriately chosen equivalent circuits. Their measured frequency responses agree well with theoretical predictions, and show low insertion losses in spite of their small sizes.

In this paper, we propose a new error feedback (EF) structure for 2-D separable-denominator digital filters described by a rational transfer function. In implementing two-dimensional separable-denominator digital filters, the minimum delay elements structures are common. In the proposed structure, the filter feedback-loop corresponding to denominator polynomial is placed at a different location compared to the commonly used structures. The proposed structure can minimize the roundoff noise more than the previous structure though the number of multipliers is less than that of previous one. Finally, we present a numerical example by designing the EF on the proposed structure and demonstrate the effectiveness of the proposed method.

In this paper, we propose a new on-line adaptive relaxation algorithm for an inverse filter in a multichannel sound reproduction system. The fluctuation of room transfer functions degrades reproduced sound in conventional sound reproduction systems in which the coefficients of the inverse filter are fixed. In order to resolve this problem, an iterative relaxation algorithm for an inverse filter performed by truncated singular value decomposition (adaptive TSVD) has been proposed. However, it is difficult to apply this method within the time duration of the sound of speech or music in the original signals. Therefore, we extend adaptive TSVD to an on-line-type algorithm based on the observed signal at only one control point, normalizing the observed signal with the original sound. The result of the simulation using real environmental data reveals that the proposed method can always carry out the relaxation process against acoustic fluctuation, for any time duration. Also, subjective evaluation in the real acoustic environment indicates that the sound quality improves without degrading the localization.

This study presents a method of realizing second order band-pass filters with planar inductive π-network. The proposed filter is more flexible in practical implementation than those using magnetic or electric coupling methods. Electromagnetic simulation results show that the bandwidth of the filter is quite insensitive to the variation in substrate thicknesses and physical layout. A 5.2 GHz filter prototype is designed and fabricated. The measured insertion loss is less than 2.3 dB in the designed pass band and the attenuations at the stop bands are all greater than 30 dB.

Periodically nonuniform coupled microstrip line (PNC-ML) loaded with transverse slits is characterized using the fullwave method of moments and short-open calibration technique. Guided-wave characteristics of both even- and odd-modes are thoroughly investigated in terms of two extracted per-unit-length transmission parameters, i.e., phase constants and characteristic impedances. As such, frequency-dependent coupling between the lines of the finite-extended PNCML is exposed via two dissimilar impedances. Meanwhile, two phase constants try to be equalized at a certain frequency by properly adjusting the slit depth and periodicity, aiming at realizing the transmission zero. Further, equivalent J-inverter network parameters of this finite-length PNCML are derived to reveal the relationship between the transmission zero and harmonic resonance. By allocating this zero to the frequency twice the fundamental passband, one-stage and two-stage PNCML filters are then designed, fabricated and measured to showcase the advantageous capacity of the proposed technique in harmonic suppression.

This paper proposes a genetic programming method to synthesize passive filter circuits. This method allows both the circuit topology and the component values to be evolved simultaneously. Experiments show that this method is fast and capable of generating circuits which are more economical than those generated by traditional design approaches. In addition, we take into account practical design considerations at high-frequency applications, where the component values are frequency-dependent and restricted to some discrete values. Experimental results show that our method can effectively generate not only compliant but also economical circuits for practical design tasks.

In this paper, we propose a circuit configuration for the low-frequency second-order active RC BPF (band pass filter) which has stable high Q. This proposed circuit is a high Q low-frequency one with a small capacitance, which is realized by applying an output capacitance multiplier to the circuit. Then a detailed circuit analysis is performed for the proposed circuit. From the simulation results of fo and Q for various combinations of circuit element values, we can confirm that the circuit realization of a center frequency of several Hz is possible by employing chip condensers of dozens of nF. The bread-board circuit of this configuration is confirmed to have small temperature dependences of fo and Q by the experiment. It is also clarified from detailed noise analysis and noise measurement that the circuit noise is sufficiently maintained at a low level.

A high-speed transconductance-C-opamp integrator using a current-feedback amplifier is proposed. The integrator has good frequency response compared with a conventional transconductance-C-opamp integrator using a voltage-feedback amplifier. The current-feedback amplifier shifts the second pole of the proposed integrator to the upper frequency. The frequency is proportional to the current gain of the current-feedback amplifier. The proposed integrator can eliminate effects of the parasitics at the output node of the transconductance since the voltage at the node is fixed. One of the circuit examples of the proposed integrator is shown. Its validity is confirmed through HSPICE simulations. The proposed integrator works as predicted up to 260 MHz.

Based on the periodical-loaded principle, a new wider stop-band filter is presented. The design equations are provided, the validity of which is proved by the measured results. Compared with loaded stub of length 1/4λg, the improved T-shape stub can change admittance paralleled with microstrip line and widen the band width of the band-stop filter. The size of the filter loaded by one side can be reduced by 2/3. The stop-band filter loaded by one side and two sides are simulated and realized. The filter loaded by two sides can achieve very wide stop-band. In addition, the stop-band of the new type of filter is deep and steep.

This paper proposes a new angular measurement system to a moving target in the presence of clutter. We apply MUSIC (MUltiple SIgnal Classification) to the outputs of a Doppler filter bank consisting of quadrature mirror filter (QMF). The comparison between QMF and the short time Fourier transform (STFT) as a preprocessor of MUSIC is also discussed. DOA estimation performance by QMF-MUSIC is nearly equal to that of STFT-MUSIC. On the other hand, QMF-MUSIC overcomes STFT-MUSIC in the aspect of computational cost. In a specific example in this paper, the proposal QMF bank by Daubechies (4th order) wavelet requires 80% fewer the number of multiplications and 25% fewer the number of additions than the FFT-based STFT filter bank.

Press-induced long-period fiber gratings exhibiting strong core-to-cladding mode coupling were formed in photonic crystal fiber. Only one resonance peak was observed over a 600 nm spectral range and the resonant wavelength was tuned over the whole range by tilting a groove plate before pressing the fiber. The resonant wavelength decreased with increasing periodicity of the grating, which was opposite to the trend of the step-index conventional optical fiber. Meanwhile, the resonant wavelength increased with increasing the ambient refractive index, which was also opposite to that of the conventional optical fiber.

This paper reports on the suitability of the SUSAN filter for the removal of artifacts that result from quantization errors in wavelet video coding. In this paper two extensions of the original filter are described. The first uses a combination of 2-D spatial filtering followed by 1-D temporal filtering along motion trajectories, while the second extension is a pure 3-D motion compensated SUSAN filter. The SUSAN approach effectively reduces coding artifacts, while preserving the original signal structure, by relying on a simple pixel-difference-based classification procedure. Results reported in the paper clearly indicate that both extensions efficiently reduce ringing that is the prevalent artifact perceived in wavelet-based coded video. Experimental results indicate an increase in perceptual as well as objective (PSNR) decoded video quality, which is competitive with state-of-the-art post-processing algorithms, especially when low computational demands of the proposed approach are taken into account.