Archives of Acoustics

This paper proposes a noise reduction unit for internal combustion engine exhaust systems based on acoustic metamaterials. To address the limitations of traditional mufflers, including large volume, high exhaust resistance, and poor low-frequency noise suppression, an acoustic model with a ring structure and variable refractive index regions was developed using equivalent medium theory. Phase control is achieved through helically wound acoustic channels. Numerical simulations of sound field distribution, transmission loss, and flow characteristics demonstrate effective low-frequency noise attenuation and stable performance across a wide frequency range. The structure achieves transmission loss exceeding 10 dB over 85% of the 500 Hz to 1000 Hz band and maintains performance at higher frequencies. Experimental validation confirms that combining the metamaterial unit with conventional mufflers enhances targeted noise reduction while maintaining acceptable exhaust resistance. The proposed design offers a compact and efficient solution for engine noise control.

In the sound field simulation of cabin-sized enclosures, the Schroeder frequency (SF) is still employed to estimate the crossover frequency (CF) that determines the validity ranges of wave-based and geometrical acoustic methods. However, because cabin-sized enclosures exhibit distinct modal behaviors from typical medium- and large-scale rooms, the validity of SF in such enclosures has not been thoroughly tested. This study introduces the modal density-based crossover frequency (MDCF) to systematically evaluate the applicability of SF in cabin-sized enclosures. The MDCF employs the same dense modal criterion as SF. However, its modal parameters, are derived from the numerical eigenfrequency analysis. This contrasts with the SF formula, where these parameters are determined solely by the room volume and reverberation time. Ten models are constructed for evaluation, grouped into two volume sets: 8m3 (cabin-sized) and 80m3 (common-sized). Each set comprises five distinct geometrical shapes from rectangular models to simplified vehicle shapes. The results reveal that, for cabin-sized enclosures under low absorption boundary conditions, the MDCF is typically 70 Hz to 150 Hz lower than SF; the discrepancies decrease to 20 Hz to 50 Hz in 80m3 rooms. Furthermore, the MDCF varies with room shapes at a constant volume, while the SF remains nearly unchanged. These findings demonstrate that MDCF provides a more reliable CF estimation for rooms with irregular shapes, highlighting the importance of considering accurate modal parameters in the acoustic analysis of cabin-sized models.

Due to the complexity of the infrasound environment and the high costs associated with data collection, frequent acquisition of infrasound data is often impractical, resulting in a limited amount of labeled data. To address the challenge of low classification prediction accuracy caused by data scarcity, this paper proposes an infrasound prediction model based on a time-series generative adversarial network (TimeGAN) and coordinated attention prototype network (CAPN) (TimeGAN-CAPN). The model begins by introducing TimeGAN, where the generative network is trained using a combination of unsupervised and supervised learning. This approach enables the network to operate within the latent space of temporal features and generate time-series data that closely aligns with the distribution of the original data. These generated samples are then combined with the original data to form an augmented dataset. Subsequently, the augmented data is input into the CAPN, which enhances the sample size per class, allowing for more precise class prototypes and improving the prediction accuracy of the model. Furthermore, the quality and diversity of the data generated by TimeGAN are quantitatively and qualitatively assessed using maximum mean discrepancy (MMD) and t-distributed stochastic neighbor embedding (t-SNE), facilitating a comparison and verification of the generated data’s performance. Experimental results show that TimeGAN-CAPN significantly outperforms the CAPN model in classification tasks with limited infrasound data, achieving an increase in accuracy of 7.15 %. This demonstrates that the proposed method is highly effective for predicting infrasound-related disasters, particularly in scenarios with small sample sizes.

A conventional cone loudspeaker has a limited capacity for creating the impression of spatiality, while a distributed mode loudspeaker (DML) has an inherent ability to evoke it. DMLs have their specific drawbacks, but some of these can be compensated for. A key question arises – is it a cone loudspeaker or a compensated DML that is preferred by listeners? A listening experiment with carefully controlled conditions was carried out to answer this question; 30 subjects participated. The participants evaluated three stereo systems: one based on a DML speaker (with its power response equalized) and two conventional two-way active systems. Two perceptual attributes were evaluated: overall preference and spatial impression. A graded pairwise comparison was used as an experimental paradigm; the results were analyzed according to the law of comparative judgment. The findings indicated that, even though the DMLs achieved slightly lower ratings than the conventional systems on average, the perceptual differences were very small. This was confirmed by the hypothesis testing that was performed on the raw results of the pairwise comparisons.

This study investigates relationships between facial, head, and neck dimensions and vocal acoustic parameters in Polish speakers. A total of 111 adults (including 30 males) participated in voice recordings of sustained vowels and anthropometric measurements. Statistical analyses using Pearson’s correlation and multiple regression revealed significant associations between anatomical features and acoustic parameters in both sexes. In males, larger head circumference, wider faces, and higher noses were linked to lower formants and more stable voices. In females, larger head and neck circumferences corresponded to lower formants and greater voice stability, while wider facial structures were associated with reduced jitter and longer phonation time. These findings may support applications in biometric identification and forensic voice analysis.

This study introduces a proof-of-concept methodology for evaluating pressure-dependent non-linear acoustic properties of liver tissue. The proposed non-linearity index (NLI) is derived from echo amplitudes obtained at two substantially different acoustic pressures. Unlike previous harmonic-based approaches, the method relies solely on the fundamental frequency band, allowing clinical implementation without additional system modifications. The NLI is estimated as a ratio of local amplitudes of the amplified low-pressure image (ALPI) to the high-pressure image (HPI). Experimental validation includes hydrophone measurements, k-Wave simulations, and in vivo imaging of healthy and fatty livers. The results demonstrate that NLI values increase with the degree of steatosis, in agreement with theoretical expectations based on tissue B/A coefficients.

This paper investigates and demonstrates the effects of three significant environmental contributors: temperature, depth and salinity impact on the acoustic signal propagation across distinctive ocean layers: mixed, thermocline, and deep layers. In the field of underwater wireless sensor networks (UWSN), exact and precise determination of coordinates for sensor localization is very crucial for data validation. Temperature dominates the upper layers; depth becomes the prime factor for the deeper domain with minimal thermal variations. Salinity while having a diminished effect, facilitates minor changes in propagation and deviation of acoustic signal speed. In our work we have analyzed these interdependencies by using different empirical models (e.g., Mckenzie, Medwin) customized to each layer, accounting to their incomparable environmental parameters. In mixed layers, changes in sound speed are mainly caused by thermal factors, where depth is of minimal importance, and the influence of salinity is insignificant, but with increasing depth, the temperature begins to decrease, and depth (pressure) begins to become important, and changes in salinity and temperature become almost equivalent. By evaluating ocean layer specified empirical formulas, we have calculated the average speed of sound and measure the corresponding contribution of all parameters. Our work has provided a substructure which helps to optimize the identification or localization of UWSN nodes. The results of this work underscored the essential to have an adaptive sound speed modeling in order to achieve enhanced and precise acoustic signal communication systems.

Active sonar signal matching is a critical technique for measuring inter-signal similarity and enhancing target detection and classification performance. However, in complex underwater environments, noise, reverberation, and prolonged signal durations often degrade matching accuracy and computational efficiency. To address these challenges, this paper proposes an adaptive extremum-aligned boundary-constrained dynamic time warping (AEB-DTW) algorithm, based on the classical dynamic time warping (DTW) framework. The algorithm extracts significant extrema from signal envelopes to suppress noise and reverberation while capturing salient features. By integrating the position and amplitude of extrema, an adaptive weighted matching strategy is introduced to enhance feature discrimination. In addition, spline fitting is applied to the residuals of the extremum matching path to dynamically generate upper and lower boundary constraints, thus restricting DTW computation to a meaningful region and achieving a balance between accuracy and efficiency. Experiments using lake-trial active sonar data under signal-to-reverberation ratios (SRRs) from 0 dB to 30 dB show that AEB-DTW outperforms Euclidean distance (ED), DTW, and its variants in matching accuracy, robustness, and angular resolution, while significantly improving computational efficiency, particularly for long-duration signals.

To reduce the size and enhance the efficiency of cascaded sandwich transducers with conical horns, a novel structural configuration of such transducers is investigated. This transducer incorporates two sets of piezoelectric stacks, enabling two-stage amplification to improve efficiency. An equivalent circuit model of the cascaded sandwich transducer with a conical horn is established, and analytical expressions for key performance parameters, including input impedance, velocity amplification ratio, and resonant characteristics, are systematically derived. Through theoretical and simulation analyses, the dynamic influence of key structural parameters on electromechanical energy conversion efficiency is determined, specifically including the output radius of the second stage, the relative position of the variable cross-sections of two sets of piezoelectric ceramic sandwich structures, and the spacing between the two sets of piezoelectric stacks. Furthermore, a performance optimization strategy based on piezoelectric single-crystal materials is proposed. Numerical simulation results, validated against the theoretical models, reveal the governing principles of piezoelectric material properties on transducer performance. Experimental results demonstrate excellent agreement between the operational characteristics of the optimized transducer and predictions obtained from both theoretical models and finite element simulations. This work provides significant guidance for the optimization of multi-mode transducers and demonstrates promising application potential in high-power ultrasonic fields.

Acoustic scattering scale models often fail to meet the acoustic similarity design requirements due to limitations in fabrication technology, testing facilities, and safe transportation, which restrict the accurate extrapolation of acoustic scattering characteristics between scaled models and full-scale ships. To overcome this challenge, the present study applies highlight model theory to perform acoustic similarity analysis and to correct the local target strength of simple objects in the model based on overall acoustic scattering correction. A novel method for correcting the target strength of non-proportional scaled models is proposed. The method is validated using various model geometries, including ellipsoids, finite-length cylinders, truncated elliptical cones, and complex structures. Additionally, the plate element method is employed for target strength correction and scaling conversion analysis for non-proportional scaled models. The study highlights the variation in target strength due to changes in geometric dimensions and demonstrates the effectiveness of the proposed correction method. The results indicate that the proposed correction approach allows for more accurate extrapolation of target strength from non-proportional scaled models to full-scale prototypes, thereby better satisfying the requirements of practical engineering applications.

The research described in this article concerns sound-insulating enclosures used for sound sources imitating a noisy machine or device. It is a continuation of experimental tests and modelling studies previously conducted on a prototype test stand, in which the enclosure walls measured 0.7 m × 0.7 m. The main aim of the current research was to estimate the acoustic efficiency of the enclosures through experimental testing on a new stand with walls measuring 0.55 m × 0.55 m, conducted under conditions similar to those found in an industrial facility. Tests conducted on five wall types of varying thicknesses, made of materials such as steel, aluminium, and plexiglass, enabled the development of a calculation model for insertion loss (IL), which can be used based on the material data for the enclosure walls. The model was validated through further experimental tests covering four additional material variants, and a high correlation of the results was obtained. The influence of the calculation model used for the enclosure wall transmission loss on the IL result was also investigated. The results of the experimental tests and modelling studies were also compared with those obtained for a larger enclosure made of the same wall materials. The research described in the current article may have practical applications in the selection of walls of cube-shaped enclosures and in estimating their effectiveness in a cost-free manner, assuming that appropriate material data are used in the calculations.

The article presents an active vibration damping system for a thin-walled cylindrical tube, which is a simplified model of a lightweight robot (LWR) arm. The proposed solution integrates control algorithms, piezoelectric materials and a hardware and software environment enabling real-time control. Macro fiber composite (MFC) elements were used for active vibration reduction, acting simultaneously as sensors and actuators. The object on which the research was conducted was a tube with an external diameter of 40 mm, this element was rigidly mounted at a distance of 1 meter from the free end, simulating cantilever conditions. The stimulation of the object to vibration was carried out using the MFC actuator, while the system response was recorded in the xPC Target environment. Based on the measurement data, the mathematical model of the object was identified in the discrete domain using the ARX method. The obtained model was used to design a controller based on the pole location method, which was implemented on a real test stand. The experimental results showed the effectiveness of the designed control system in reducing the amplitude of natural vibrations of the structure. The use of MFC elements as sensor elements and actuators enabled effective vibration damping in real time, confirming the usefulness of the proposed solution in the context of improving the precision of robotic systems.

Assessing the impact of flanking sound transmission is one of the most significant challenges in the process of designing building partitions. Acoustic parameters declared by manufacturers of lightweight systems are subject to errors of up to several decibels – and in the case of inaccurate construction on site, these differences can reach even higher values. One factor contributing to this is the phenomenon known as flanking sound transmission, which involves the transmission of acoustic energy through partitions connected to the partition directly dividing two adjacent rooms. For this reason, estimating the resultant acoustic insulation of a partition, taking flanking paths into account, is crucial at an early stage of the design process to ensure compliance with the requirements outlined in standard recommendations, and the literature. Currently, there are regulations and studies that provide guidance on calculating the estimated reduction in acoustic insulation due to flanking transmission. However, in practice, situations arise that have not yet been addressed in standards or the literature. Examples include partitions made of plasterboard, which are among the most common types of partition walls in Poland, yet are not covered by current normative procedures, as well as glass systems. This study aims to further explore this topic by analysing the impact of combining a massive partition with flanking lightweight partitions for selected structures (glass, plasterboard with single or double panelling, with full or partial sound-absorbing material infill, and without infill) and connection types.