In this paper, we investigated the overfitting characteristics of nonlinear equalizers based on an artificial neural network (ANN) and the Volterra series transfer function (VSTF), which were designed to compensate for optical nonlinear waveform distortion in optical fiber communication systems. Linear waveform distortion caused by, e.g., chromatic dispersion (CD) is commonly compensated by linear equalizers using digital signal processing (DSP) in digital coherent receivers. However, mitigation of nonlinear waveform distortion is considered to be one of the next important issues. An ANN-based nonlinear equalizer is one possible candidate for solving this problem. However, the risk of overfitting of ANNs is one obstacle in using the technology in practical applications. We evaluated and compared the overfitting of ANN- and conventional VSTF-based nonlinear equalizers used to compensate for optical nonlinear distortion. The equalizers were trained on repeated random bit sequences (RRBSs), while varying the length of the bit sequences. When the number of hidden-layer units of the ANN was as large as 100 or 1000, the overfitting characteristics were comparable to those of the VSTF. However, when the number of hidden-layer units was 10, which is usually enough to compensate for optical nonlinear distortion, the overfitting was weaker than that of the VSTF. Furthermore, we confirmed that even commonly used finite impulse response (FIR) filters showed overfitting to the RRBS when the length of the RRBS was equal to or shorter than the length of the tapped delay line of the filters. Conversely, when the RRBS used for the training was sufficiently longer than the tapped delay line, the overfitting could be suppressed, even when using an ANN-based nonlinear equalizer with 10 hidden-layer units.

The existing literature focuses on the applications of fast filter bank due to its excellent frequency responses with low complexity. However, the topic is not addressed related to the general transfer function expressions of the corresponding subfilters for a specific channel. To do this, in this paper, general closed-form transfer function expressions for fast filter bank are derived. Firstly, the cascaded structure of fast filter bank is modelled by a binary tree, with which the index of the subfilter at each stage within the channel can be determined. Then the transfer functions for the two outputs of a subfilter are expressed in a unified form. Finally, the general closed-form transfer functions for the channel and its corresponding subfilters are obtained by variables replacement if the prototype lowpass filters for the stages are given. Analytical results and simulations verify the general expressions. With such closed-form expressions lend themselves easily to analysis and direct computation of the transfer functions and the frequency responses without the structure graph.

This paper discusses non-ideal issues in a fully passive noise shaping successive approximation register analog-to-digital converter. The fully passive noise shaping techniques are realized by switches and capacitors without operational amplifiers to be scalable and power efficient. However, some non-ideal issues, such as parasitic capacitance, comparator noise, thermal noise, will affect the performance of the noise shaping and then degrade the final achievable resolution. This paper analyzes the effects of the main non-ideal issues and provides the design reference for fully passive noise shaping techniques. The analysis is based on 2nd order fully passive noise shaping SAR ADC with an 8-bit architecture and an OSR of 4.

An improved radiometric calibration algorithm by extending the Mitsunaga and Nayar least-square minimization based algorithm with two major ideas is presented. First, a noise & outlier removal procedure based on the analysis of brightness transfer function is included for improving the algorithm's capability on handling noise and outlier in least-square estimation. Second, an alternative minimization formulation based on weighted least square is proposed to improve the weakness of least square minimization when dealing with biased distribution observations. The performance of the proposed algorithm with regards to two baseline algorithms is demonstrated, i.e. the classical least square based algorithm proposed by Mitsunaga and Nayar and the state-of-the-art rank minimization based algorithm proposed by Lee et al. From the results, the proposed algorithm outperforms both baseline algorithms on both the synthetic dataset and the dataset of real-world images.

This letter proposes an extended image-source model to simulate the room transfer function for a rotatable complex source in a three-dimensional reverberant room. The proposed model uses spherical harmonic decomposition to describe the exterior sound field from the complex source. Based on “axis flip” concept, the mirroring relations between the source and images are summarized by a unified mirroring operator that occurs on the soundfield coefficients. The rotation movement of the source is taken into account by exploiting the rotation property of spherical harmonics. The accuracy of our proposed model is verified through the appropriate simulation examples.

This paper reports on the relationships between the performance of retro-reflectors and the sharpness of an aerial image formed with aerial imaging by retro-reflection (AIRR). We have measured the retro-reflector divergence angle and evaluated aerial image sharpness by use of the contrast-transfer function. It is found that the divergence angle of the retro-reflected light is strongly related to the sharpness of the aerial image formed with AIRR.

A single-dimensional interface which enables users to obtain diverse localizations of audio sources is proposed. In many conventional interfaces for arranging audio sources, there are multiple arrangement parameters, some of which allow users to control positions of audio sources. However, it is difficult for users who are unfamiliar with these systems to optimize the arrangement parameters since the number of possible settings is huge. We propose a simple, single-dimensional interface for adjusting arrangement parameters, allowing users to sample several diverse audio source arrangements and easily find their preferred auditory localizations. To select subsets of arrangement parameters from all of the possible choices, auditory-localization space vectors (ASVs) are defined to represent the auditory localization of each arrangement parameter. By selecting subsets of ASVs which are approximately orthogonal, we can choose arrangement parameters which will produce diverse auditory localizations. Experimental evaluations were conducted using music composed of three audio sources. Subjective evaluations confirmed that novice users can obtain diverse localizations using the proposed interface.

This paper proposes an opamp-free solution to implement single-phase-clock controlled noise shaping in a SAR ADC. Unlike a conventional noise shaping SAR ADC, the proposal realizes noise shaping by charge redistribution, which is a passive technique. The passive implementation has high power efficiency. Meanwhile, since the proposal maintains the basic architecture and operation method of a traditional SAR ADC, it retains all the advantages of a SAR ADC. Furthermore, noise shaping helps to improve the performance of SAR ADC and relaxes its non-ideal effects. Designed in a 65-nm CMOS technology, the prototype realizes 58-dB SNDR based on an 8-bit C-DAC at 50-MS/s sampling frequency. It consumes 120.7-µW power from a 0.8-V supply and achieves a FoM of 14.8-fJ per conversion step.

This paper analyzes three passive noise shaping techniques in a SAR ADC. These passive noise shaping techniques can realize 1st and 2nd order noise shaping. These proposed opamp-less noise shaping techniques are realized by charge-redistribution. This means that the proposals maintain the basic architecture and operation principle of a charge-redistribution SAR ADC. Since the proposed techniques work in a passive mode, the proposals have high power efficiency. Meanwhile, the proposed noise shaping SAR ADCs are robust to feature size scaling and power supply reduction. Flicker noise is not introduced into the ADC by passive noise shaping techniques. Therefore, no additional calibration techniques for flicker noise are required. The noise shaping effects of the 1st and 2nd order noise shaping are verified by behavioral simulation results. The relationship between resolution improvement and oversampling rate is also explored in this paper.

This paper proposes an on-chip measurement method of PLL through fully digital interface. For the measurement of the PLL transfer function, we modulated the phase of the PLL input in triangular form using Digital-to-Time Converter (DTC) and read out the response by Time-to-Digital Converter (TDC). Combination of the DTC and TDC can obtain the transfer function of the PLL both in the magnitude domain and the phase domain. Since the DTC and TDC can be controlled and observed by digital signals, the measurement can be conducted without any high speed analog signal. Moreover, since the DTC and TDC can be designed symmetrically, the measurement method is robust against Process, Voltage, and Temperature (PVT) variations. At the same time, the employment of the TDC also enables a measurement of the PLL lock range by changing the division ratio of the divider. Two time domain circuits were designed using 180nm CMOS process and the HSPICE simulation results demonstrated the measurement of the transfer function and lock range.

This paper proposes a method based on modulation transfer function (MTF) to restore the power envelope of noisy reverberant speech by using a Kalman filter with linear prediction (LP). Its advantage is that it can simultaneously suppress the effects of noise and reverberation by restoring the smeared MTF without measuring room impulse responses. This scheme has two processes: power envelope subtraction and power envelope inverse filtering. In the subtraction process, the statistical properties of observation noise and driving noise for power envelope are investigated for the criteria of the Kalman filter which requires noise to be white and Gaussian. Furthermore, LP coefficients drastically affect the Kalman filter performance, and a method is developed for deriving LP coefficients from noisy reverberant speech. In the dereverberation process, an inverse filtering method is applied to remove the effects of reverberation. Objective experiments were conducted under various noisy reverberant conditions to evaluate how well the proposed Kalman filtering method based on MTF improves the signal-to-error ratio (SER) and correlation between restored power envelopes compared with conventional methods. Results showed that the proposed Kalman filtering method based on MTF can improve SER and correlation more than conventional methods.

Binaural reproduction is one of the promising approaches to present a highly realistic virtual auditory space to a listener. Generally, binaural signals are reproduced using a set of headphones that leads to a simple implementation of such a system. In contrast, binaural signals can be presented to a listener using a technique called “transaural reproduction” which employs a few loudspeakers with crosstalk cancellation for compensating acoustic transmissions from the loudspeakers to both ears of the listener. The major advantage of transaural reproduction is that a listener is able to experience binaural reproduction without wearing any device. This leads to a more natural listening environment. However, in transaural reproduction, the listener is required to be still within a very narrow sweet spot because the crosstalk canceller is very sensitive to the listener's head position and orientation. To solve this problem, dynamic transaural systems have been developed by utilizing contact type head tracking. This paper introduces the development of a dynamic transaural system with non-contact head tracking which releases the listener from any attachment, thereby preserving the advantage of transaural reproduction. Experimental results revealed that sound images presented in the horizontal and median planes were localized more accurately when the system tracked the listener's head rotation than when the listeners did not rotate their heads or when the system did not track the listener's head rotation. These results demonstrate that the system works effectively and correctly with the listener's head rotation.

Sensing and reproduction of precise sound-space information is important to realize highly realistic audio communications. This study was conducted to realize high-precision sensors of 3D sound-space information for transmission to distant places and for preservation of sound data for the future. Proposed method comprises a compact and spherical object with numerous microphones. Each recorded signal from multiple microphones that are uniformly distributed on the sphere is simply weighted and summed to synthesize signals to be presented to a listener's left and right ears. The calculated signals are presented binaurally via ordinary binaural systems such as headphones. Moreover, the weight can be changed according to a human's 3D head movement. A human's 3D head movement is well known to be a crucially important factor to facilitate human spatial hearing. For accurate spatial hearing, 3D sound-space information is acquired as accurately reflecting the listener's head movement. We named the proposed method SENZI (Symmetrical object with ENchased ZIllion microphones). The results of computer simulations demonstrate that our proposed SENZI outperforms a conventional method (binaural Ambisonics). It can sense 3D sound-space with high precision over a wide frequency range.

A method to estimate sound source orientation in a reverberant room using a microphone array is proposed. We extend the conventional modeling of a room transfer function based on the image method in order to take into account the directivity of a sound source. With this extension, a transfer function between a sound source and a listener (or a microphone) is described by the superposition of transfer functions from each image source to the listener multiplied by the source directivity; thus, the sound source orientation can be estimated by analyzing how the image sources are distributed (power distribution of image sources) from observed signals. We applied eigenvalue analysis to the spatial correlation matrix of the microphone array observation to obtain the power distribution of image sources. Bsed on the assumption that the spatial correlation matrix for each set of source position and orientation is known a priori, the variation of the eigenspace can be modeled. By comparing the eigenspace of observed signals and that of pre-learned models, we estimated the sound source orientation. Through experiments using seven microphones, the sound source orientation was estimated with high accuracy by increasing the reverberation time of a room.

This work presents a four-channel headset achieving a 5.1-channel-like hearing experience using a low-complexity head-related transfer function (HRTF) model and a simplified reverberator. The proposed down-mixing architecture enhances the sound localization capability of a headset using the HRTF and by simulating multiple sound reflections in a room using Moorer's reverberator. Since the HRTF has large memory and computation requirements, the common-acoustical-pole and zero (CAPZ) model can be used to reshape the lower-order HRTF model. From a power consumption viewpoint, the CAPZ model reduces computation complexity by approximately 40%. The subjective listening tests in this study shows that the proposed four-channel headset performs much better than stereo headphones. On the other hand, the four-channel headset that can be implemented by off-the-shelf components preserves the privacy with low cost.

A design methodology for implementation of low-noise switched-capacitor low-pass filter (SC LPF) with small capacitance spread is proposed. The proposed method is focused on the reduction of operational amplifier noise transfer gain at low frequencies and the reduction of total capacitance. A new SC LPF topology is proposed in order to adapt the correlated double sampling and charge scaling technique at the same time. Design examples show that proposed filter reduces the total capacitance by 65% or more compared to the conventional one without having significant increase in noise transfer gain.

This paper presents analysis and design of passive RC polyphase filters (RCPFs) in tutorial style. Single-phase model of a single-stage RCPF is derived, and then, multi-stage RCPFs are analyzed and obtained some restrictions for realizable poles and zeros locations of RCPFs. Exact design methods of RCPFs with equal ripple type, and Butterworth type responses are explained for transfer function design and element value design along with some design examples.

This paper presents a new common-mode stabilization method for a CMOS differential cascode Class-E power amplifier with LC-tank based driver stage. The stabilization method is based on the identification of the poles and zeros of the closed-loop transfer function at a critical node. By adding a series resistor at the common-gate node of the cascode transistor, the right-half-plane poles are moved to the left half plane, improving the common-mode stability. The simulation results show that the new method is an effective way to stabilize the PA.

A technique to reduce noise transfer functions (NTF) of switched-capacitor (SC) integrators without changing their signal transfer functions (STF) is proposed. The proposed technique based on a simple reconnection scheme of multiple sampling capacitors. It can be implemented into any SC integrators as long as they have a transfer delay. A design strategy is also given to reduce the effect of parasitic capacitors. An SC integrator with a small total capacitance and a low noise transfer gain based on the proposed technique is also proposed. For a given design example, the total capacitance and the simulated noise transfer gain of the proposed SC integrator are 37% and 90% less than the conventional one.

This paper presents an accurate and systematic method to simulate the interference imposed on the input/output (I/O) ports of electronic equipment under the electrical fast transients/burst (EFT/B) test. The equivalent circuit of the EFT/B generator and the coupling clamp are modeled respectively. Firstly, a transfer function (TF) of the EFT pulse-forming network is constructed with the latent parameters based on circuit theory. In the TF, two negative real parameters characterize the non-oscillation process of the network while one complex conjugate pair characterizes the damping-oscillation process. The TF of the pulse-forming network is therefore synthesized in the equivalent circuit of the EFT/B generator. Secondly, the standard coupling clamp is modeled based on the scatter (S) parameter obtained by using a vector network analyzer. By applying the vector fitting method during the rational function approximation, a macromodel of the coupling clamp can be obtained and converted to a Spice compatible equivalent circuit. Based on the aforementioned procedures, the interference imposed on the I/O ports can be simulated. The modeling methods are validated experimentally, where the interference in differential mode and common mode is evaluated respectively.