Applied Acoustics最新文献

{"title":"Sound source localization and detection based on densely connected network and attention mechanism","authors":"Bomao Zhou,&nbsp;Jin Tang","doi":"10.1016/j.apacoust.2024.110338","DOIUrl":"10.1016/j.apacoust.2024.110338","url":null,"abstract":"<div><div>Sound Source Localization and Detection (SSLD) is a joint task of detecting sound event activity and sound source localization (SSL). This paper proposes an SSLD method based on dense connection and attention mechanism. We propose a densely connected block parallel to the gated linear unit (GLU) to enhance feature propagation and alleviate vanishing gradient problems. We introduce multi-headed self-attention (MHSA) to aggregate contextual information and model long-term dependencies. The output adopts activity-coupled Cartesian direction-of-arrival (ACCDOA) representation. We decomposed the loss function based on Euclidean distance and proposed a loss function based on vector length loss and vector angle loss. These two types of losses correspond to sound activity detection (SAD) and direction-of-arrival (DOA) estimation in the SSLD task, respectively. The experimental results indicate that the proposed model achieves state-of-the-art performance compared to the baseline models. The proposed loss function also achieved performance improvement compared to the general Euclidean distance loss. In addition, we propose a system for remote SSL. In a real-world environment, we used two Laser Doppler vibrometers (LDVs) to remotely capture sound signals from different types of target objects for remote SSL. The results of real-world experiments have validated the effectiveness of the proposed system and demonstrated its potential application value.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-13","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142419013","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"An efficient joint training model for monaural noisy-reverberant speech recognition","authors":"Xiaoyu Lian,&nbsp;Nan Xia,&nbsp;Gaole Dai,&nbsp;Hongqin Yang","doi":"10.1016/j.apacoust.2024.110322","DOIUrl":"10.1016/j.apacoust.2024.110322","url":null,"abstract":"<div><div>Noise and reverberation can seriously reduce speech quality and intelligibility, affecting the performance of downstream speech recognition tasks. This paper constructs a joint training speech recognition network for speech recognition in monaural noisy-reverberant environments. In the speech enhancement model, a complex-valued channel and temporal-frequency attention (CCTFA) are integrated to focus on the key features of the complex spectrum. Then the CCTFA network (CCTFANet) is constructed to reduce the influence of noise and reverberation. In the speech recognition model, an element-wise linear attention (EWLA) module is proposed to linearize the attention complexity and reduce the number of parameters and computations required for the attention module. Then the EWLA Conformer (EWLAC) is constructed as an efficient end-to-end speech recognition model. On the open source dataset, joint training of CCTFANet with EWLAC reduces the CER by 3.27%. Compared to other speech recognition models, EWLAC maintains CER while achieving much lower parameter count, computational overhead, and higher inference speed.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-13","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142419015","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"A square cross-section impedance tube for extending the effective frequency range","authors":"Yong-Bin Zhang,&nbsp;Zu-Jie Yang,&nbsp;De-Shan Li,&nbsp;Jia-Zhu Li,&nbsp;Xiao-Zheng Zhang,&nbsp;Chuan-Xing Bi","doi":"10.1016/j.apacoust.2024.110342","DOIUrl":"10.1016/j.apacoust.2024.110342","url":null,"abstract":"<div><div>The most widely used method for measuring the acoustic properties of materials is the two-microphone method (TMM). However, the TMM is limited by the plane wave assumption, which constrains its effective frequency range. While simultaneously utilizing an impedance tube with smaller diameter can expand this range, it also intensifies edge constraint effects and increases measurement inconvenience. To address these issues, this paper constructs a square cross-section impedance tube and applies the modal cancellation method (MCM) to extend the effective frequency range. The proposed method can extract plane waves within a cut-off frequency of order (2, 2) by summing sound pressures measured by four microphones positioned at the midpoint of each side of a cross section of a square tube. As a result, it is capable of measuring acoustic properties across a frequency range three times wider than that achievable with conventional TMM, with a more straightforward implementation and without the need to reduce tube size. Finally, simulations and experiments are conducted to verify the validity of proposed method.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142419014","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Rolling bearing remaining useful life prediction method based on vibration signal and mechanism model","authors":"Xiuliang Zhao ,&nbsp;Ye Yang ,&nbsp;Qian Huang ,&nbsp;Qiang Fu ,&nbsp;Ruochen Wang ,&nbsp;Limei Wang","doi":"10.1016/j.apacoust.2024.110334","DOIUrl":"10.1016/j.apacoust.2024.110334","url":null,"abstract":"<div><div>Variations in operating conditions and usage environments, bearings often exhibit multiple degradation modes (DMs). Existing degradation models fail to inadequately capture the various degradation trends of bearings, resulting in low accuracy in predicting the remaining useful life (RUL). To address this challenge, a RUL prediction method based on the vibration signal and a mechanism model is proposed. Firstly, signal processing and feature fusion techniques are employed to construct a nonlinear composite health indicator (CHI) as a measure of bearing life degradation. Then, an adaptive degradation starting point (DSP) identification method is employed to segment the bearing’s full life cycle into distinct states. Based on the results of state division, DMs are classified into three types, including the abrupt DM, the progressive DM and the self-healing DM. Secondly, the degradation mechanisms of bearings are analyzed, and a mechanism model is proposed to describe multiple DMs simultaneously. This model is compared with typical life degradation models, demonstrating superior adaptability to the self-healing DM. Finally, a novel RUL prediction method is developed based on the proposed mechanism model. This method allows for predicting the RUL of bearings under various DMs. Compared to other prediction methods, the proposed approach reduces the mean absolute relative error by at least 58.72% and improves the score by at least 363.21%.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142419062","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Effects of spring water sounds on psychophysiological responses in college Students: An EEG study","authors":"Nan Zhang ,&nbsp;Yuqin Zhang ,&nbsp;Fengcheng Jiao ,&nbsp;Chao Liu ,&nbsp;Jing Shi ,&nbsp;Weijun Gao","doi":"10.1016/j.apacoust.2024.110318","DOIUrl":"10.1016/j.apacoust.2024.110318","url":null,"abstract":"<div><div>While the beneficial effects of spring water sound on health are recognized, the specific psychophysiological mechanisms at varying sound pressure levels (SPLs) remain unclear. This study explores the effects of different SPLs of spring water sound on psychophysiological responses, utilizing subjective evaluations alongside electrocardiogram (ECG) and electroencephalogram (EEG) monitoring. Thirty-eight college students were exposed to five SPLs of spring water sounds (40, 45, 50, 55, and 60 dBA). Participants underwent subjective evaluation using the Profile of Mood States (POMS) and Restoration Outcome Scale (ROS), complemented by physiological measurements through ECG and EEG. Spring water sounds significantly improved psychophysiological health. Psychologically, compared to no sound, POMS scores decreased by 21.60–67.34 %, ROS scores increased by 1.15–1.86 times, and emotional disorders and subjective stress were significantly improved (p &lt; 0.05). Physiologically, the low frequency/high frequency (LF/HF) ratio of ECG decreased by 2.30–30.48 % for 40–50 dBA levels, and the overall EEG power for 50–55 dBA levels decreased by 12.72–21.83 %. Significant enhancements in relative power (RP) of α and β (24.21–73.34 % and 23.96–85.26 %, respectively) and a reduction in mental stress (11.23–59.60 %) were observed. The 50 dBA spring water sound notably influenced brain order–disorder balance, showing a 2.01–3.32-fold improvement over other SPLs. A significant physiological inflection point occurred at 1.0 min post-exposure across all measures. Strong correlations were found between ΔPOMS and ΔRP α (r = -0.535, p &lt; 0.01), and between ΔROS and ΔOverall power (r = -0.418, p &lt; 0.001). This study provides a method to comprehensively analyze the positive effects of water sound from both psychological and physiological perspectives and offers a reference for the design of sound environments.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-10","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142418924","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Design of adjustable low-frequency sound absorbers through hybrid digital-analog shunt loudspeakers","authors":"Yubing Xu,&nbsp;Chaonan Cong","doi":"10.1016/j.apacoust.2024.110331","DOIUrl":"10.1016/j.apacoust.2024.110331","url":null,"abstract":"<div><div>Low frequency noise is the primary frequency range that affects individuals’ physical and emotional well-being. The impact of conventional low-frequency sound absorption technologies is restricted by spatial constraints. The shunt loudspeaker is an emerging technology for controlling low-frequency noise, where the acoustic impedance can be modified by a shunt circuit. However, the imprecision and instability caused by the nonlinear impact of loudspeakers and the parasitic resistance of electronic components in analog circuits is challenging. To overcome the problem, this research proposes an adjustable low-frequency sound absorber based on a hybrid digital-analog shunt loudspeaker. The equivalent circuit model is initially formulated, and the design of the adjustable synthetic impedance is then developed. Simulations and experiments are conducted to validate the theoretical model. The proposed absorber demonstrates excellent capabilities in absorbing low-frequency sound with greater precision, stability, independent of the cavity depth and ease of adjustment, enabling a wide range of practical applications.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142418922","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Generation of bessel-like beam by a binary ultrasonic lens","authors":"Huan Wang,&nbsp;Min He,&nbsp;Jiajie He,&nbsp;Xue Jiang,&nbsp;Ying Li,&nbsp;Dean Ta","doi":"10.1016/j.apacoust.2024.110320","DOIUrl":"10.1016/j.apacoust.2024.110320","url":null,"abstract":"<div><div>Bessel beams have already been used in acoustic tweezers, ultrasonic medical imaging, and Doppler velocity estimation due to their non-diffractive and self-healing properties. We proposed a binary ultrasonic lens, with ultra-thin planar periodic structures for efficient conversion of the plane wave into a Bessel-like beam. The effectiveness of long-axis focusing has been demonstrated through simulations and experiments with FWHM of 8.5 mm and 9.5 mm around 10 mm. By varying the operating frequency of the incident wave from 1.9 MHz to 2.2 MHz, the position of the long-axis focusing point can be adjusted experimentally from 23.05 mm to 28.95 mm. Additionally, the beam generated by the binary ultrasonic lens can form a focused sound field when passing through multi-layered tissues and the self-healing capability of the Bessel-like beam has been numerically validated, showing strong robustness. This binary ultrasonic lens for a Bessel-like beam would conform better to ergonomic and miniaturization requirements, and expand versatile applications in clinical diagnosis and therapy.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142418920","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Fan noise radial mode detection based on L1/2 regularization method","authors":"Baohong Bai","doi":"10.1016/j.apacoust.2024.110340","DOIUrl":"10.1016/j.apacoust.2024.110340","url":null,"abstract":"<div><div>Radial mode detection has been widely employed in fan noise research. Many radial mode detection algorithms by conducting the convex <span><math><mrow><msub><mi>L</mi><mn>1</mn></msub></mrow></math></span> norm optimization have been proposed by taking advantage of the fact that the fan noise sound field is usually dominated by a limited set of radial modes. Nevertheless, <span><math><mrow><msub><mi>L</mi><mn>1</mn></msub></mrow></math></span> norm optimization has been shown to be suboptimal and cannot enforce further sparsity. A fan noise radial mode detection method based on <span><math><mrow><msub><mi>L</mi><mrow><mn>1</mn><mo>/</mo><mn>2</mn></mrow></msub></mrow></math></span> regularization algorithm is proposed in this paper to improve its robustness and dynamic range. This method employs the <span><math><mrow><msub><mi>L</mi><mrow><mn>1</mn><mo>/</mo><mn>2</mn></mrow></msub></mrow></math></span> regularization to directly solve the in-duct sound propagation model. Synthesized simulations indicates that the <span><math><mrow><msub><mi>L</mi><mrow><mn>1</mn><mo>/</mo><mn>2</mn></mrow></msub></mrow></math></span> regularization method can detect the correct radial mode order and amplitude under low signal-to-noise ratio. Moreover, the new method is insensitive to regularization parameter. An experimental system of radial rake is implemented to validate the new method. It is shown that the L-curve method, which do not require any prior information of the acoustic mode distribution or contaminating measurement noise, is an appropriate choice of the regularization parameter determination method for the <span><math><mrow><msub><mi>L</mi><mrow><mn>1</mn><mo>/</mo><mn>2</mn></mrow></msub></mrow></math></span> regularizer. It is demonstrated that the new <span><math><mrow><msub><mi>L</mi><mrow><mn>1</mn><mo>/</mo><mn>2</mn></mrow></msub></mrow></math></span> regularization method can enhance the mode sparsity, especially under low signal-to-noise ratio. Moreover, the resolution and dynamic range can be improved by the <span><math><mrow><msub><mi>L</mi><mrow><mn>1</mn><mo>/</mo><mn>2</mn></mrow></msub></mrow></math></span> regularization method, which makes it an effective and robust fan noise radial mode detection technique in practical applications.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142418921","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"An integrated system for real-time monitoring of temperature changes, cavitation and necrosis inside the tissue at the focus of the HIFU beam using a backscattered ultrasonic signals inducing changes in the electrical power driving the HIFU transducer","authors":"ChengLong Hu,&nbsp;Qi Cao,&nbsp;Xin Ai,&nbsp;Zi Yi Chen,&nbsp;Yanhao Li","doi":"10.1016/j.apacoust.2024.110325","DOIUrl":"10.1016/j.apacoust.2024.110325","url":null,"abstract":"<div><div>To address the unclear mechanism of changes in the electrical power driving the HIFU transducer and the challenges in real-time monitoring of treated tissue damage, this paper presents a system for real-time monitoring of driving electric power, focus of the HIFU beam, acoustic cavitation, temperature detection, and necrotic lesion monitoring using voltage and current sensors, cavitation-detecting probes, and temperature sensors. Utilizing the characteristics of backscattered ultrasonic signals, we investigated the mechanism and pattern of changes in the electrical power of the HIFU transducer and examined the relationship between the electrical power driving the HIFU transducer and the temperature, cavitation, and acoustic impedance of the treated tissue. Experimental results demonstrated that the system accurately monitors the output electrical power during treatment, characterizes the intensity of cavitation activity by the amplitude of driving power during sonication, and identifies necrotic lesions by changes in the amplitude of backscattered signals detected in the time interval between sonications. The system can distinguish types of transient changes in the physical properties of the treated tissue in the region of the HIFU beam focus, which is intended to increase the precision of HIFU sonication dose regulation and ensure the safety of treatment. Additionally, it provides a solution for real-time monitoring of cavitation activity and necrotic lesion severity in treated tissue during HIFU treatment.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142418918","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"A modified active noise control algorithm for convex combined structures based on partial updating of weight matrices","authors":"Jingqing Jia ,&nbsp;Shuming Chen ,&nbsp;Yongnan Zhao ,&nbsp;Feng Qian","doi":"10.1016/j.apacoust.2024.110333","DOIUrl":"10.1016/j.apacoust.2024.110333","url":null,"abstract":"<div><div>Convex combination schemes with fast tracking ability and good steady-state restoration performance have attracted wide attention in Active Noise Control (ANC) systems. The scheme can combine the outputs of multiple filters. However, the time domain filter-X least mean square (FxLMS) algorithm used in traditional convex combination structure has the problem of high computational complexity. Compared with the conventional FxLMS algorithm, the time–frequency domain FxLMS (TF-FxLMS) algorithm can compute gradient estimates and filter reference signals in the frequency domain to reduce the computation. Since the algorithm is a block processing of the signal in the frequency domain, the presence of the fast fourier transform (FFT) introduces unwanted delays which causes the rate of convergence to slow down. FFT operations should be performed at a sampling interval, which limits the real-time implementation of the algorithm. In order to reduce the computational complexity of traditional convex combination and improve the performance of the algorithm, a convex combined structure algorithm of normalized filter-X least mean square (NFxLMS) algorithm and TF-FxLMS algorithm is proposed in this paper. In addition, in order to further reduce the amount of computation, the NFxLMS algorithm is decomposed into NFxLMS algorithm and TF-FxLMS algorithm by means of partial updating of weight matrix. In this paper, the computational complexity of the proposed algorithm is analyzed and compared with the traditional algorithm. The simulation results show that the proposed algorithm has better noise reduction effect than the traditional algorithm under the noise input conditions of various driving conditions. At the same time, it has the advantage of lower computational complexity.</div></div>","PeriodicalId":55506,"journal":{"name":"Applied Acoustics","volume":null,"pages":null},"PeriodicalIF":3.4,"publicationDate":"2024-10-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"142418917","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":2,"RegionCategory":"物理与天体物理","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}