This letter theoretically analyzes and minimizes the L2-sensitivity for all-pass fractional delay digital filters of which structure is given by the normalized lattice structure. The L2-sensitivity is well known as one of the useful evaluation functions for measuring the performance degradation caused by quantizing filter coefficients into finite number of bits. This letter deals with two cases: L2-sensitivity minimization problem with scaling constraint, and the one without scaling constraint. It is proved that, in both of these two cases, any all-pass fractional delay digital filter with the normalized lattice structure becomes an optimal structure that analytically minimizes the L2-sensitivity.

This paper proposes a method of watermarking for digital audio signals based on adaptive phase modulation. Audio signals are usually non-stationary, i.e., their own characteristics are time-variant. The features for watermarking are usually not selected by combining the principle of variability, which affects the performance of the whole watermarking system. The proposed method embeds a watermark into an audio signal by adaptively modulating its phase with the watermark using IIR all-pass filters. The frequency location of the pole-zero of an IIR all-pass filter that characterizes the transfer function of the filter is adapted on the basis of signal power distribution on sub-bands in a magnitude spectrum domain. The pole-zero locations are adapted so that the phase modulation produces slight distortion in watermarked signals to achieve the best sound quality. The experimental results show that the proposed method could embed inaudible watermarks into various kinds of audio signals and correctly detect watermarks without the aid of original signals. A reasonable trade-off between inaudibility and robustness could be obtained by balancing the phase modulation scheme. The proposed method can embed a watermark into audio signals up to 100 bits per second with 99% accuracy and 6 bits per second with 94.3% accuracy in the cases of no attack and attacks, respectively.

In this paper, we propose a rectangular weighting function that can be used in the method of iteratively reweighted least squares (IRWLS) for designing equiripple all-pass IIR filters. The purpose of introducing this weighting function is to improve the convergence performance in the solution of the IRWLS. The height of each rectangle is designed to be equal to the local maximum of each ripple, and the width of each rectangle is designed so that the area of each rectangle becomes equal to the area of each ripple. Here, the ripple is the absolute value of the phase error. We show experimentally that the convergence performance in the solution of the IRWLS can be improved by using the proposed weighting function.

This letter proposes an analog active noise control (ANC) circuit with an all-pass filter (APF). To improve performance of the previously reported analog ANC circuit, we inserted an APF to the circuit in order to fit phases of a noise and an electrical signal in the circuit. As a result, we confirmed improvement of the noise canceling effect of the analog ANC circuit.

Development of an artificial reverberator for low-memory requirements is an issue of importance in applications such as mobile multimedia devices. One possibility is to use an All-Pass Filter (APF), which is embedded in the feedback loop of the comb filter network. In this paper, we propose a reverberator employing time-varying APFs to increase the reverberation performance. By changing the gain of the APF, we can increase the number of frequency peaks perceptually. Thus, the resulting reverberation sounds much more natural, even with less memory, than the conventional approach. In this paper, we perform theoretical and perceptual analyses of artificial reverberators employing time-varying APF. Through the analyses, we derive the degree of phase variation of the APF that is perceptually acceptable. Based on the analyses, we propose a method of designing artificial reverberators associated with the time-varying APFs. Through subjective tests, it is shown that the proposed method is capable of providing perceptually comparable sound quality to the conventional methods even though it uses less memory.

Howling is very annoying problem to the hearing-aid users and it limits the maximum usable gain of hearing aids. We propose a new feedback cancellation system by inserting a time-varying decorrelation filter in the forward path. We use a second-order all-pass filter with control parameters whose time variation is implemented using a low-frequency modulator. A noticeable reduction of weight-vector misalignment is achievable using our proposed method.

The optimal design of complex infinite impulse response (IIR) two-channel quadrature mirror filter (QMF) banks with equiripple frequency response is considered. The design problem is appropriately formulated to result in a simple optimization problem. Therefore, based on a variant of Karmarkar's algorithm, we can efficiently solve the optimization problem through a frequency sampling and iterative approximation method to find the complex coefficients for the IIR QMFs. The effectiveness of the proposed technique is to form an appropriate Chebyshev approximation of a desired response and then find its solution from a linear subspace in several iterations. Finally, simulation results are presented for illustration and comparison.

This paper introduces a way to realize high-pass, band-stop and all-pass transfer functions using two-integrator loop structure consisting of loss-less and lossy integrators. The basic circuit configuration is constructed with five Operational Transconductance Amplifiers (OTAs) and two grounded capacitors. It is shown that the circuit can realize their circuit transfer functions by choosing the input terminals, and that the circuit parameters can also be independently set by the transconductance gains with the proportional block. Although the basic circuit configuration has been known, it seems that the feature for realizing the high-pass, the band-stop and the all-pass transfer functions makes the structure more attractive and useful. An example is given together with simulated results by PSPICE.

Analog inverter is one of the most useful building blocks in analog circuits. This paper proposes an analog inverter consisting of a p-channel MOS (PMOS) and an n-channel MOS (NMOS) inverter and presents an application to all-pass filter realizations. The proposed circuit has a wide dynamic range by combining PMOS and NMOS inverters. When the proposed analog inverter is applied to an all-pass filter, the circuit configuration becomes simpler and occupies less chip area and power consumption.

This paper is devoted to a new design method for infinite impulse response approximate inverse system of a nonminimum phase system. The design is carried out such that the convolution of the nonminimum phase polynomial and its approximate inverse system can be represented by an approximately linear phase all-pass filter. A method for estimating the time delay and order of an approximate inverse system is also presented. Using infinite impulse response approximate inverse systems better accuracy is achieved with reduced computational complexity. Numerical examples are included to show the effectiveness of the proposed method.

This letter presents a new design method for approximate inverse systems using all-pass networks. The efficacy of approximate inverse systems for input and parameter estimation of nonminimum phase systems is well recognized. in the previous methods, only time domain design of FIR (finite impulse response) type approximate inverse systems were considered. Here, we demonstrate that IIR (infinite impulse response) type approximate inverse systems outperform the previous methods. A nonlinear optimization technique is adopted for designing the proposed system in the frequency domain. Numerical examples are also presented to show the effectiveness of the proposed method.

In this paper we present a design method for all-pass networks with consideration of the stability. It is based on the eigen filter method and Remez exchange algorithm is used to obtain the equiripple phase error solution. In the iteration of the proposed algorithm, the eigen values besides maximum eigen value are used in order to obtain a stable all-pass networks.