Room acoustics is mainly based on the reverberation theories of Sabine and Eyring. In Sabine's theory however, the reverberation time does not reach zero, even if the condition of absolute absorption is fulfilled. Eyring revised reverberation theory to resolve this contradiction. However, Eyring's theory has an inconsistency between the formulations of the steady-state and decay processes. Therefore, the author revised Sabine's theory, taking a different approach from that of Eyring. This revised theory was constructed by introducing the concept of "reverberation of a direct sound." In this study, a new mathematical model of reverberation using reflection orders is proposed. This is a reconstruction of the author's revised theory. The new model includes the temporal energy distribution in each reflection order and uses the concept of "reverberation of a direct sound" for the entire reverberation process. It shows that the concept is also essential for the reflected sounds. In addition, the reverberation decay agrees with the revised theory previously proposed by the author. Overall, the new model showed good agreement with the simulation results.

This paper is a companion to "Reexamination of the favorable reverberation time of concert halls measured in a 3D synthesized sound field," Acoust. Sci. & Tech., 45, 204–215 (2024). Anechoic music sources were reproduced by a virtual orchestra set on concert hall stages and were recorded at audience seats. Four music excerpts were chosen. By an Ambisonics playback in the laboratory, a series of psychological experiments were conducted. Twenty-one music experts judged the clarity, sound strength, spaciousness, and overall acoustical quality of the presented sound. Adding the results from the previous paper, a regression analysis on the relationships between contributing subjective attributes and objective parameters found that EDT_M and C_80,3 contributed to clarity, G_M (or G_L) and RT_M to sound strength, and BQI and G_L to spaciousness. Here, subscripts "L," "M," and "3" denote the octave band averages for 125 and 250, 500 and 1,000, and 500, 1,000 and 2,000 Hz, respectively, and "E" designates early sound, i.e., less than 80 ms. Favorable ranges of physical parameters for each subjective attribute were determined. Reverberance, spaciousness, and clarity were identified as significant subjective attributes contributing to overall acoustic quality, with the corresponding physical metrics being RT_M, G_L, and BQI.

In this paper, we propose an easily designable low-frequency acoustic metasurface (AMS) absorber composed of multiple imperfect microslit resonators designed to achieve near-perfect sound absorption within a one-third-octave-band. Some specific designs of one-third-octave-band near-perfect absorbers at 125, 250, and 500 Hz are presented. We have developed a robust and efficient user-friendly absorber design method combining the transfer matrix method and a unique geometry design rule of component resonators. To develop this design method, we conducted extensive numerical and experiment-based examinations by thermoviscous acoustic simulation and impedance tube measurements, particularly addressing the number of component resonators and their peak sound absorption coefficient. The numerical and experimental results demonstrated the importance of creating a coupled resonator with the appropriate number of imperfect component resonators, each with a lower sound absorptivity peak. These features are crucially important for achieving thin sound absorbers without compromising the desired sound absorption properties. Numerical sound absorptivity evaluation revealed that using more component resonators to create a coupled resonator enables individual component resonators to operate as resonators with lower sound absorptivity peaks. This simple operation achieves robust sound absorption characteristics with less degradation.

Single-channel blind dereverberation aims to remove reverberation from a single-channel reverberant signal without using any prior knowledge. In acoustics, weighted prediction error (WPE), a method mainly used for a multi-channel signal, is often applied for this task. However, it is difficult to achieve well-performed dereverberation for a single-channel signal. In this paper, for better single-channel dereverberation, we propose to simultaneously estimate the source signal and the room impulse response (RIR) instead of only predicting reverberation. By modeling convolution using matrix lifting in the time-frequency domain, we formulate the dereverberation problem as a non-convex optimization problem of recovering a sparse rank-1 matrix. In sparse regularization, we introduce reweighting, enabling the improvement of sparse matrix recovery. The alternating direction method of multipliers (ADMM) with acceleration is applied to approximately solve the optimization problem, resulting in closed form updates. In our experiments, we confirmed that the proposed method outperforms existing methods in several reverberant conditions and is capable of removing both early reflection and late reverberation. MATLAB code of the proposed method is available online (https://doi.org/10.24433/CO.3541617.v1).

This paper presents the current design of the J-AESOP corpus, a learner speech corpus featuring Japanese speakers' English. It has been developed as part of the Asian English Speech cOrpus Project (AESOP), an international and multi-institutional project to construct a collection of Asian English speech databases. While the recording procedures and speech materials are standardized in the AESOP project, the J-AESOP corpus incorporates additional features not found in other AESOP corpora, such as data from native English speakers, Japanese reading materials (Japanese version of "The North Wind and the Sun"), manual correction of automatic forced alignment, and perceptual ratings of accentedness/nativelikeness and comprehensibility. These unique features allow an in-depth investigation of Japanese-English bilingual speech, as exemplified by our exploratory investigation of the production of voiceless coronal fricatives in Japanese (i.e., [s, ɕ]) and English (i.e., /s, ∫, θ/) reported in this paper. The paper also discusses directions for further development of the corpus, including improvements in data availability.

This study introduces two efficient methods for selecting Tikhonov regularization parameters in acoustical inverse problems. The first approach employs a binary search (BS) algorithm to identify the regularization parameter that satisfies a predefined power constraint. Compared to traditional iterative searches over N candidate values, BS reduces the number of iterations from N to log ₂N. The second method, Adaptive Normalized Tikhonov (ANT), combines the conventional L-curve and Normalized Tikhonov techniques. By fitting the ratio of the inverse system matrix's largest eigenvalue to an exponential decay function during preprocessing at a few sample frequencies, ANT determines the regularization parameter with a single calculation for other frequencies. Both methods were experimentally validated in a multi-zone sound field reproduction scenario using a measured reverberant room impulse responses database. Results demonstrated that BS achieves a balance between reproduction accuracy and robustness while significantly improving efficiency. The ANT method provided the most stable system without iterative calculations. These improvements indicate that both approaches offer compelling solutions for real-time applications.

This study explores the design and implementation of an IIR (Infinite Impulse Response) all-pass filter that simulates cochlear delay characteristics. This paper contains three topics: filter design, implementation, and musical evaluation. First, we designed an IIR all-pass filter to simulate cochlear delay characteristics by optimizing its zeros and poles to achieve the desired group delay. Additionally, the filter was implemented as a VST (Virtual Studio Technology) plug-in for real-time applications and is publicity available. Next, subjective evaluations were conducted to assess the musical impact of this filter. We applied the filter to snare drum, bass drum, bass guitar, and electric guitar to explore its musical applicability. Participants compared the filtered and original sounds. Percussion instruments received mixed feedback, with the filter sometimes described as "artificial." In contrast, string instruments like bass guitar and electric guitar were rated as "impressive" and "attractive," suggesting greater relevance for these sounds. Finally, we investigated the impact of the filter on guitar performance. Performance deviations from a metronome were measured under 10 different conditions by varying the number of filters and delay times. The results indicated that excessive delay introduced by the filter could disrupt synchronization during performances.

In general, a speech signal can be measured by a microphone, such as a throat microphone. However, the speech signal measured by a microphone often contains surrounding noise. On the other hand, although a throat microphone is effective for surrounding noise, the speech signal it measures includes body-conducted internal noise. In this study, we propose an improvement method for the sound quality of the speech signal measured by a throat microphone to achieve speech recognition well. The relationship between the original speech signal and the speech measured by the throat microphone is not clear. Therefore, we consider the relationship as a multiplicative and additive model of the original speech signal and noise components with unknown parameters. An algorithm is proposed to simultaneously estimate the original speech signal and the unknown parameters using Bayes' theorem based on the speech signal measured by the throat microphone. Finally, a speech recognition experiment is conducted to confirm the effectiveness of the proposed algorithm.

We construct an orthonormal basis for interior problems of the Helmholtz equation, based on the properties of a reproducing kernel Hilbert space defined by the spectral characteristics of interior sound fields. The constructed basis coincides with what is commonly known as spherical basis functions. Furthermore, leveraging the structure of this space, we derive the addition theorem in a compact form. This facilitates the conversion between reproducing kernel representations and spherical harmonic expansions and provides insights into estimating spherical harmonic coefficients from sampled measurements.

In this study, we measured the acoustic characteristics of nonwood baseball bats modified according to the Revised Japanese Product Standards (hereafter, Safe Goods (SG) Standards) enforced in 2024. New standard bats showed peak frequencies approximately 500 Hz higher than previous models. During Spring Koshien 2024, players reported differences in bat sound and ball travel distance, with the onomatopoeic description changing from "kakkin" to "kyu-in" following the revision, according to various media. The results of acoustic measurements conducted in compliance with the SG Standards confirm the observations of the players, indicating a tonal shift in the bats after the SG Standards were revised.

In this study we compare the prosody of Japanese with that of Māori and New Zealand English, the contact language, as impressionistic analysis of Māori and Japanese indicates prosodic similarities, despite many other differences. This may be due to the fact that proto-Japanese just like Māori stems from the Pacific region. However, Māori under the influence of English changed substantially. As an indirect way of comparing the prosody we devised a perception experiment using delexicalized speech employing Japanese listeners. Most listeners were able to differentiate between Japanese and NZ English, but did not place Māori closer to Japanese than English.