In this paper, we propose a metaporous absorber with air-filled resonators that enhances low-frequency sound absorption for thin porous materials with low flow resistivity and a simple theoretical model under the assumption of plane wave propagation inside the material. The metaporous absorber features a periodic array structure made up of unit cells, within which microslit resonators are strategically placed in the porous material. Three distinct unit cells, each exhibiting unique sound absorption characteristics, are proposed. Combining two of these unit cells makes it possible to offset the shortcomings of each other's sound absorption capabilities, resulting in a broader range of highsound-absorption effects. Firstly, a transfer matrix modeling of the metaporous absorbers is proposed, numerically verified by the finite element method and experimentally validated by impedance tube measurement. Using the constructed transfer matrix model and genetic algorithm optimization, we designed two highly efficient near-perfect sound absorbers at frequencies from 700 Hz to 1.7 kHz and experimentally demonstrated their sound absorption characteristics. The present absorber is particularly effective for enhancing the performance of thin porous materials with lower flow resistivity.
Pulsed pressure waves propagating through a narrow air gap surrounded by spherical walls of different diameters were computationally and experimentally investigated to generate an impact force for noncontact and nondestructive testing. It was confirmed that with the proposed model, an impulse-like wave with peak positive pressure exceeding 10 kPa can be obtained with a gap width of 0.9 mm at a laser energy of 35 mJ.
A cellular polypropylene film metallized with gold electrodes is directly glued to the back plate of a commercially available condenser microphone cartridge whose metallic diaphragm is removed, composed of a microphone set with an existing pre-amplifier and power module. Lack of vibration resonance of the diaphragm makes it possible to widen frequency ranges in pressure sensitivity compared with those of commonly used condenser microphones. In fact, it has been experimentally verified that the prototyped microphone has good response to sound waves over the extremely wide range of frequencies from 200 Hz to 400 kHz and is highly acceptable to intense sound pressures of up to several tens of kilo-pascals.
This study investigates strategies to enhance the loading and dispensing capabilities of a droplet in ultrasonic levitation systems through acoustic field optimization. Using a 28.58 kHz transducer, two approaches were evaluated: (1) horizontal standing waves with a 5 degrees angled reflector at first-to third-order resonances, and (2) inclined standing waves at different angles (first-order resonance) under a fixed surface vibration velocity of 0.8 m/s. Results show that the horizontal configurations with the angled reflector required up to 112.5% higher surface vibration velocity to reach comparable levitation performance to that of parallel reflectors, revealing inefficiencies in reflector-angle adjustments. In contrast, tilting the standing wave angle to 45 degrees significantly enhanced stability, enabling reliable levitation of an averaged 1.0 mL droplet with reduced energy input. The inclined-wave method outperformed reflector-angle modifications, achieving precise droplet insertion and dispensing while minimizing acoustic energy consumption.
Environmental noise poses significant challenges, necessitating effective sound insulation. Sonic crystals (SCs) have the potential to become a building material that provides both sound insulation and ventilation but suffers from a narrow sonic bandgap (SBG) and anisotropic behavior. In this study, sound insulation properties of two-dimensional triangular-lattice hierarchical sonic crystals (HSCs), in which a hierarchical structure is applied to SCs, were examined by the finite element method. Results indicate that multiple lattice constants facilitate broader SBGs, with second-order SBGs exhibiting isotropy. This suggests that HSCs can mitigate the incident angle dependence.
Liquid crystal (LC) varifocal lenses are characterized by their need for compactness and high-speed response, rendering them well-suited for next-generation optical devices. The focal length can be modulated by reorienting LC molecules through acoustic radiation force. In this study, the influence of the geometrical structure of ultrasonic LC lenses on the optical performance was examined. Two LC lenses with distinct glass substrate thicknesses were fabricated, and their optical characteristics were evaluated. The electro-mechanical parameters were found to be altered by the thickness of the glass substrate, which consequently led to an improvement in the power consumption in focus tunability.
Recent studies have demonstrated that combinations of logarithmic durations can classify duration-sensitive phonemes, such as Japanese singleton and geminate consonants, at various speaking rates. The acoustic features of the Japanese fricative /s/ and affricate /ts/ are related to duration; therefore, a combination of logarithmic durations can likely classify these consonants. To examine this possibility, discriminant models using linear and logarithmic durations, with and without a speaking-rate-related variable, were compared in terms of their performance in classifying /s/ and /ts/ at word-initial position at various speaking rates. The results indicate that the discriminant model using logarithmic duration with a speaking-rate-related variable can classify the consonants better than the other models, indicating the importance of logarithmic duration. The results are considered in the framework of logarithmic information processing in the brain.
In general, the resonance frequencies of ultrasonic emitters that use bolt-clamped Langevin transducers differ slightly. However, when using these emitters in an arrayed device, the resonance frequencies of each emitter must be matched. In this paper, the arbitrary reduction of the resonance frequency by adding a small amount of mass after fabrication is examined. The resonance frequency was lowered by adding more mass, demonstrating the utility of this method.
A method for adjusting room acoustics through active control using conventional loudspeakers as secondary sources is proposed to suppress inhomogeneity caused by standing waves and room modes. Unlike conventional active control, the control target is the specific acoustic impedance, which is the ratio of sound pressure to particle velocity. The aim is to approach a sound field with only direct waves and no reflections. First, to perform the control, the condition for the maximum absorption coefficient of the virtual boundary surface was determined from the perspective of impedance matching, and an error function was set. Next, a particle velocity measurement method was introduced to obtain the values of specific acoustic impedance, and the weighting of sound pressure and particle velocity was modified. Furthermore, using these tools, experiments were conducted using both a real sound field and simulations to verify the effect of impedance control. Finally, impedance control was reinterpreted from the perspective of microphone directivity, clarifying the control mechanism. The results confirm the method's effectiveness in low-frequency sound field adjustment, where passive absorbers are insufficient.
This study investigated whether individuals with musical experience could accurately perceive changes in the singing voice and underlying vocal tract movements of a singer who completed one year of vocal training, using only the singing voice. Vocal tract modifications were analyzed using real-time magnetic resonance imaging and evaluated by professional singers, instrumentalists, and students. Professional singers demonstrated more nuanced evaluations of vocal tract shape. Instrumentalists, while capable of assessing voice quality, showed less differentiation across vocal tract features. Students used narrower rating ranges and struggled to assess both aspects. These findings indicate that musical background influences evaluative tendencies regarding voice quality and vocal tract configurations.
Auditory spatial attention is crucial for extracting relevant sounds from background noise in noisy environments. Despite its significance in daily life, the effect of auditory spatial attention on the depth direction remains poorly understood. The present study aimed to investigate how auditory selective attention influences the detection of target sounds in the depth direction using sensitivity (d'), false alarm rates, and reaction time (RT) for the target sound. In each trial, either a target or distractor sound was presented from one of the five distances (32, 64, 96, 128, and 160 cm). The listeners were directed to respond as soon as they heard the target sound, while ignoring distractor sounds. The results indicated that directing attention to a specific distance significantly increased the sensitivity (d') at that distance compared to other distances. Furthermore, the false alarm rate was the lowest at the attended position and progressively increased as sound positions deviated from the focus of attention. However, no significant effect of attention on the RT was observed. These findings suggest that auditory selective attention is not limited to the horizontal direction but can also operate along the depth direction in reverberant environments, expanding our understanding of auditory spatial attention.
To bypass phase estimation, complex-valued generative models have been developed to directly handle spectra of audio signals. The complex-valued restricted Boltzmann machine (CRBM) is one of such promising models proposed recently. However, similar to the other models, CRBM cannot treat the logarithmic nature of auditory perception important to realize a better model for audio application. This is because CRBM handles complex values in the rectangular coordinate (i.e., real and imaginary parts), which hinders applying the logarithmic transform to magnitude. To overcome this drawback of CRBM, we propose the gamma-von-Mises (GVM) RBM that models complex-valued spectra in the polar coordinate (i.e., magnitude and phase). GVM RBM handles magnitude by the gamma distribution using the logarithmic function and phase by the von Mises distribution. Our objective and subjective experiments showed that GVM RBM outperformed the other models including CRBM and complex-valued variational autoencoder (CVAE).
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.
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 log2 N. The second method, Adaptive Normalized Tikhonov (ANT), combines the conventional Lcurve 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.
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.
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.
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, c]) and English (i.e., /s, f, 0/) reported in this paper. The paper also discusses directions for further development of the corpus, including improvements in data availability.
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 nearperfect 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.
