Sound Source Identification Research Articles

Experiment on Noise Source Identification Based on Acoustic Array

In order to effectively control the noise source, the experiments on sound source identification were conducted on the generator side, intake side, and exhaust side of the test bench of the 2VQS EFI Volkwagen engine under different working conditions using the acoustic array. And applying the beamforming algorithm, the main noise sources at frequency range of 1000∼2000Hz were located at the three test surfaces. The result show that: In front of generator side, the source was mainly located at the generator, the crankshaft pulley and the intake manifold and the strongest sound pressure was focused on the generator; In front of intake side, the source was mainly located at the generator, the cylinder body, and the transmission case, and the strongest sound pressure was focused on the cylinder body; In front of exhaust side, the source was mainly located at the cylinder body, the three-way catalytic converter and the transmission case, and the strongest sound pressure was focused on the generator pulley and gear system. The strongest sound pressure of the three test surfaces increase with the rotational speed of engine, was nearly equal to 2500r/min, and changed smoothly more than 2500r/min.It provided a way for controlling engine noise.

Discrete state space method and modal extension method based impact sound synthesis model** Project supported by the National Natural Science Foundation of China (Grant Nos. 11574249 and 11874303) and the Natural Science Basic Research Plan in Shaanxi Province of China (Grant No. 2018JQ1001).

The efficient and accurate synthesis of physical parameter-controllable impact sounds is essential for sound source identification. In this study, an impact sound synthesis model of a cylinder is proposed based on discrete state space (DSS) method and modal extension method (MEM). This model is comprised of the whole three processes of the physical interaction, i.e., the Hertz contact process, the transient structural response process, and the sound radiation process. Firstly, the modal expanded DSS equations of the contact system are constructed and the transient structural response of the cylinder is obtained. Then the impact sound of the cylinder is acquired using improved discrete Raleigh integral. Finally, the proposed model is verified by comparing with existing models. The results show that the proposed impact sound synthesis model is more accurate and efficient than the existing methods and easy to be extended to the impact sound synthesis of other structures.

Array processing for the localisation of noise sources in hot flows

Abstract This paper investigates the problem of localizing a sound source in a heated flow using a microphone array. Applications are found in studies dealing with the identification of sound sources in hot turbulent jets, or with the sound radiation from installed turbofans. Two configurations have been investigated: a shear layer flow (wind-tunnel type) and a jet flow. In the present study acoustic data are generated using a simulation based on the Linearized Euler Equations. For heated flows, refraction by temperature gradients is superimposed with refraction by velocity gradients, and the objective of this study is to assess whether this effect is important and how it can be accounted for in different source localisation methods. For this purpose, a time-reversal-based imaging method has been compared with a beamforming-based method in which the time-delays are computed based on ray tracing. For the shear flow, the results show that for high subsonic Mach numbers and steep thermal gradients, the thermal stratification must be taken into account to ensure a satisfactory precision of localisation for both methods. However, including the gradients of velocity and temperature is less crucial for imaging sound sources in the jet flow. The results indicate also that the localisation error is lower with the beamforming and ray-tracing technique than with the time-reversal technique, the latter being more sensitive to the limited array aperture.

Directional Sound Detection by Sensing Acoustic Flow

The ability to determine the direction of propagation of an acoustic wave is crucial to sound source detection, localization, and identification. Current techniques based on sensing acoustic pressure using membranes at multiple spatial locations tend to have directionality and output sensitivity that depend strongly on the sound frequency, which compromises performance, especially in miniaturized applications. Unlike pressure, which is a scalar, the fluctuating flow of the acoustic medium (i.e., the acoustic particle velocity) is a vector, which inherently contains the direction information of sound propagation. Here, we experimentally demonstrate an alternative, and more direct, method of directional sound detection by sensing acoustic flow using nanofibers, which are driven by viscous forces applied by the fluctuating medium. The measured output signal of this prototype sensor exhibits bidirectional response and flat frequency response across a wide range of frequencies from 100 ${\rm Hz}$ to 10000 ${\rm Hz}$ . While pressure-sensing microphones now enjoy an annual market in the billions of units sold, a broadband sound flow sensor with almost full fidelity will provide considerable benefits.

Localization and Spatial Discrimination in Children and Adolescents with Moderate Sensorineural Hearing Loss Tested without Their Hearing Aids

The present study investigated two measures of spatial acoustic perception in children and adolescents with sensorineural hearing loss (SNHL) tested without their hearing aids and compared it to age-matched controls. Auditory localization was quantified by means of a sound source identification task and auditory spatial discrimination acuity by measuring minimum audible angles (MAA). Both low- and high-frequency noise bursts were employed in the tests to separately address spatial auditory processing based on interaural time and intensity differences. In SNHL children, localization (hit accuracy) was significantly reduced compared to normal-hearing children and intraindividual variability (dispersion) considerably increased. Given the respective impairments, the performance based on interaural time differences (low frequencies) was still better than that based on intensity differences (high frequencies). For MAA, age-matched comparisons yielded not only increased MAA values in SNHL children, but also no decrease with increasing age compared to normal-hearing children. Deficits in MAA were most apparent in the frontal azimuth. Thus, children with SNHL do not seem to benefit from frontal positions of the sound sources as do normal-hearing children. The results give an indication that the processing of spatial cues in SNHL children is restricted, which could also imply problems regarding speech understanding in challenging hearing situations.

Filter-and-sum Beamforming Sound Source Identification Algorithm for Spherical Microphone Arrays Based on Pressure Contrition

Filter-and-sum Beamforming Sound Source Identification Algorithm for Spherical Microphone Arrays Based on Pressure Contrition

Sound & vibration control for a single-cylinder gasoline engine based on parameter optimization of timing-chain system

In order to promote the idling noise quality of a single-cylinder gasoline engine, this paper addresses sound source identification and noise control research. The noise was identified by the application of subjective evaluation, acoustic spectrum and sound intensity analysis. It was found that the noise was caused by the anomalous dynamic performance of the timing system under idling conditions. Furthermore, sound and vibration characteristics of timing system were improved by design methodology research of key components. A multi-body dynamic model was established to characterize dynamic characteristics of the timing system under idling conditions. The key factor of producing noise was that the fluctuation of contact force between the chain and guide and transverse displacement of the chain were much higher than those of the allowable design limit. For the lowest design alternation and manufacturing costs, the work analyzed six timing system improvement schemes obtained by cross combination of tensioner blade line and guide strip radian parameters. After that, the optimal design scheme which could improve dynamic performance parameters of the timing system was derived. The design scheme was conducted with a acoustic test of engine to derive the following results. The noise level of a single-cylinder engine under idling conditions decreased by 3 dB(A). The abnormal noise of the original engine was eliminated under subjective evaluation. The sound quality under other working conditions had no apparent deterioration. Research shows that guide and tensioner blade line optimization design could improve dynamic performance of the timing chain system to eliminate abnormal noise, thereby significantly improving the acoustic characteristic of a single-cylinder engine.

Identification and contribution analysis of vehicle interior noise based on acoustic array technology

This article proposes an advanced far-field acoustic hologram for vehicle sound source identification and investigates a method of performing contribution analyses in the far field. The hologram is applied for statistically optimized near-field acoustic holography used to identify vehicle sound sources (internal and external). An advanced far-field acoustic hologram is first proposed to improve the spatial resolution of low-frequency sound source locations in the far field. In addition, an acoustic contribution analysis based on near-field acoustic holography is used to provide a direction for low-noise optimization design. Next, a new sound source identification and low-noise technology is explored via the aforementioned methods and applied for sound source identification and acoustic contribution analyses for a vehicle. The analysis results showed that in the process of sound source identification, the contribution of each sound source can be simultaneously sorted, and this finding can help guide research on the low-noise control of a vehicle.

Periodic boundary based FFT-FISTA for sound source identification

Compared with the conventional beamforming, the Fourier-based fast iterative shrinkage thresholding algorithm (FFT-FISTA) can effectively improve the spatial resolution and suppress the sidelobe. To furtherly achieve higher computational efficiency and better sound source identification performance, an alternative periodic boundary is utilized to replace the zero boundary of Fourier transform, a periodic boundary based FFT-FISTA is proposed in this paper. And its superiority is demonstrated by the simulation and validation experiment of equal and unequal intensity sources.

A monotonic two-step iterative shrinkage/thresholding algorithm for sound source identification based on equivalent source method

Near-field acoustical holography (NAH) based on equivalent source method (ESM) is an efficient technique applied for sound source identification. However, the conventional ESM with ℓ2 norm regularization cannot produce a satisfactory solution in high frequency, where the average array inter-element spacing is larger than half a wavelength. Therefore, conventional ESM with Tikhonov regularization is restricted to relatively low frequency. To overcome the issue, an alternative method called monotonic two-step iterative shrinkage/thresholding algorithm for near-field acoustical holography is proposed. In the algorithm, another existing algorithm called Wideband Acoustical Holography (WBH) is used to generate the threshold, and also be used as a benchmark for comparison. Simulated measurements based on an irregular microphone array are performed to compare the performances of the Tikhonov regularization, WBH and MTwIST. The results suggested that the proposed method can identify the sound sources more accurately than Tikhonov regularization in full frequency range, and it performs better than WBH in the case of identifying coherent sources at relatively low frequency. The experiments demonstrated the validity and the practicability of the proposed method.

How does activity affect soundscape assessments? Insights from an urban soundscape intervention with music

The relationship between activity and soundscape has recently garnered research attention, particularly in public spaces. In the summer of 2015, we installed an interactive sound system (Musikiosk) in a busy public park allowing users to play their own content over high-quality speakers. Questionnaires (N = 197) were administered over 3 conditions: pre-installation with park users, during the installation phase with Musikiosk users, and during the installation phase with park users not using Musikiosk. For users and observers of Musikiosk, a separate evaluation of the Musikiosk intervention was also included. The questionnaire included quantitative evaluations (soundscapes scale from Swedish Soundscape Quality Protocol, restorativeness, mood, noise sensitivity), free response data (soundscape description, self-reported activity, sound source identification, reasons for park visit), and demographic info (age, interaction with others, proximity of residence). The qualitative descriptions of activity and sound sources were categorized into emergent themes. Presented here is the analysis of the interaction between activity and soundscape assessment in terms of quantitative variables and qualitative descriptions.

Sound localization models as evaluation tools for tactical communication and protective systems.

Tactical Communication and Protective Systems (TCAPS) are hearing protection devices that sufficiently protect the listener's ears from hazardous sounds and preserve speech intelligibility. However, previous studies demonstrated that TCAPS still deteriorate the listener's situational awareness, in particular, the ability to locate sound sources. On the horizontal plane, this is mainly explained by the degradation of the acoustical cues normally preventing the listener from making front-back confusions. As part of TCAPS development and assessment, a method predicting the TCAPS-induced degradation of the sound localization capability based on electroacoustic measurements would be more suitable than time-consuming behavioral experiments. In this context, the present paper investigates two methods based on Head-Related Transfer Functions (HRTFs): a template-matching model and a three-layer neural network. They are optimized to fit human sound source identification performance in open ear condition. The methods are applied to HRTFs measured with six TCAPS, providing identification probabilities. They are compared with the results of a behavioral experiment, conducted with the same protectors, and which ranks the TCAPS by type. The neural network predicts realistic performances with earplugs, but overestimates errors with earmuffs. The template-matching model predicts human performance well, except for two particular TCAPS.

Bird Song Scene Analysis Using a Spatial-Cue-Based Probabilistic Model

[abstFig src='/00290001/22.jpg' width='300' text='Spatial-cue-based probabilistic model' ] This paper addresses bird song scene analysis based on semi-automatic annotation. Research in animal behavior, especially in birds, would be aided by automated or semi-automated systems that can localize sounds, measure their timing, and identify their sources. This is difficult to achieve in real environments, in which several birds at different locations may be singing at the same time. Analysis of recordings from the wild has usually required manual annotation. These annotations may be inaccurate or inconsistent, as they may vary within and between observers. Here we suggest a system that uses automated methods from robot audition, including sound source detection, localization, separation and identification. In robot audition, these technologies are assessed separately, but combining them has often led to poor performance in natural setting. We propose a new Spatial-Cue-Based Probabilistic Model (SCBPM) for their integration focusing on spatial information. A second problem has been that supervised machine learning methods usually require a pre-trained model, which may need a large training set of annotated labels. We have employed a semi-automatic annotation approach, in which a semi-supervised training method is deduced for a new model. This method requires much less pre-annotation. Preliminary experiments with recordings of bird songs from the wild revealed that our system outperformed the identification accuracy of a method based on conventional robot audition.**This paper is an extension of a proceeding of IROS2015.

Optimization of Fixed Microphone Array in High Speed Train Noises Identification Based on Far-Field Acoustic Holography

Acoustical holography has been widely applied for noise sources location and sound field measurement. Performance of the microphones array directly determines the sound source recognition method. Therefore, research is very important to the performance of the microphone array, its array of applications, selection, and how to design instructive. In this paper, based on acoustic holography moving sound source identification theory, the optimization method is applied in design of the microphone array, we select the main side lobe ratio and the main lobe area as the optimization objective function and then put the optimization method use in the sound source identification based on holography, and finally we designed this paper to optimize microphone array and compare the original array of equally spaced array with optimization results; by analyzing the optimization results and objectives, we get that the array can be achieved which is optimized not only to reduce the microphone but also to change objective function results, while improving the far-field acoustic holography resolving effect. Validation experiments have showed that the optimization method is suitable for high speed trains sound source identification microphone array optimization.

Who's that Knocking at My Door? Neural Bases of Sound Source Identification.

When hearing knocking on a door, a listener typically identifies both the action (forceful and repeated impacts) and the object (a thick wooden board) causing the sound. The current work studied the neural bases of sound source identification by switching listeners' attention toward these different aspects of a set of simple sounds during functional magnetic resonance imaging scanning: participants either discriminated the action or the material that caused the sounds, or they simply discriminated meaningless scrambled versions of them. Overall, discriminating action and material elicited neural activity in a left-lateralized frontoparietal network found in other studies of sound identification, wherein the inferior frontal sulcus and the ventral premotor cortex were under the control of selective attention and sensitive to task demand. More strikingly, discriminating materials elicited increased activity in cortical regions connecting auditory inputs to semantic, motor, and even visual representations, whereas discriminating actions did not increase activity in any regions. These results indicate that discriminating and identifying material requires deeper processing of the stimuli than discriminating actions. These results are consistent with previous studies suggesting that auditory perception is better suited to comprehend the actions than the objects producing sounds in the listeners' environment.

Performance Analysis and Application of Functional Beamforming Sound Source Identification

Performance Analysis and Application of Functional Beamforming Sound Source Identification

Sound source identification on automotive muffler surface using single explosion excitation and near-field acoustic holography method

Recent automobiles tend to have larger interior space, resulting in the smaller space for arranging the muffler. Flat shape mufflers are required for this reason but cause relatively large radiation noise from the surface. In order to improve understanding of radiation noise generation mechanism of flat shape muffler, we tried to identify the noise sources using the near-field acoustic holography (NAH) method on muffler surface excited by a single explosion pulse, which simulates a single pulsation in the exhaust system,. Two types of mufflers were tested with different shapes. Results show that some areas had large vibration velocity at specific frequencies depending on the muffler shape. The flat muffler tends to generate large radiation noise and vibration velocity as compared with a conventional one. In the front and rear outer plates of the muffler, large vibration velocity exists within a certain range of frequency. Within the same frequency range, the flat muffler has more number of the peak of radiation noise as compared with the conventional one.

Musical instrument identification in simulated electric acoustic hearing and in cochlear implant users with contralateral hearing aids.

While cochlear implant (CI) users achieve high levels of speech perception in quiet conditions, music perception is a challenging task. Gfeller et al. (2000) found that CI users rated music as second most important acoustic stimulus in their lives next to speech. Music is considered to be a central part of human experience, and many CI users can only dream of one day being able to appreciate music as they had before losing their hearing. For those with congenital deafness, music remains a distant hope, something experienced only through reactions of others, as indicated by poor performance of prelingually deaf children with CIs on measures of music perception (Stabej et al., 2012). Some recent work in CI field has focused on improving quality of music perception, and in doing so, improving overall quality of life for implant users (Limb & Roy, 2014; Limb & Rubinstein, 2012; Petersen, Mortensen, Hansen, & Vuust, 2012). This has posed a great challenge to researchers, as CI devices have a long history of design principles solely focused on delivering aspects of acoustic signal that lead to improved speech communication without much consideration of music perception (Loizou, 1998, 2006).A number of factors may limit music perception for CI users (McDermott, 2004). The substantial reduction in spectral resolution, limited dynamic range, and channel interactions between neighboring electrodes may render most music noisy and unpleasant to CI users (Limb & Rubinstein, 2012). Although CI users' rhythm perception is comparable with normal-hearing (NH) listeners (Gfeller & Lansing, 1991; Gfeller, Woodworth, Robin, Witt, & Knutson, 1997; Phillips-Silver et al., 2015), melody and timbre perception remain quite poor. Implant users rely heavily on rhythmic cues in identification of melodies, performing better when recognizing melodies with a distinctive rhythm than recognizing melodies with less rhythmic structure (Kong, Cruz, Jones, & Zeng, 2004; Schulz & Kerber, 1994). Their increased reliance on temporal cues may be a consequence of reduction in spectral resolution owing to a limited number of stimulating electrodes. Pitch perception has been found to vary greatly across CI users. Some listeners can discriminate a difference in pitch of one semitone reliably, whereas others require interval differences as large as two octaves for reliable judgments (Drennan & Rubinstein, 2008; Ping, Yuan, & Feng, 2012). Timbre, defined as the attribute of auditory sensation which enables a listener to judge that two nonidentical sounds, similarly presented and having same loudness and pitch, are dissimilar (ASA 111-1994) has also been found to be problematic for CI users. An important dimension of musical expressivity, timbre is primarily defined through temporal envelope and spectral shape of a sound (Dowling & Harwood, 1986; Handel, 1995), which enable identification and tracking of a musical sound source over time (McAdams & Giordano, 2016). While temporal envelopes are encoded fairly well by CIs, spectral shape is degraded (Drennan & Rubinstein, 2008). Two major limitations include less spectral resolving power (typically Timbre is an essential element in enjoying musical sound. Composers use timbre in form of different musical instruments. …

Restoring sensitivity to interaural timing differences in bilateral cochlear implant listeners using multi-electrode stimulation

Identification and discrimination of sound sources in complex auditory environments is facilitated for normal-hearing listeners from access to interaural time differences (ITDs). For patients fitted with bilateral cochlear implants (BiCI), binaural sensitivity is harder to achieve due to several factors, including asynchronous interaural processing. BiCI listeners have shown good ITD sensitivity with low rates of electrical stimulation; however, low-rate delivery of ITD cues is unrealistic for the high-rate pulsatile stimulation required to achieve good speech understanding. One solution is to present a mix of high- and low-rate stimulation on different electrodes, preserving both speech recognition and sound localization ability. In the present study, a binaural benefit was observed in a mixed-rate strategy under direct electrical stimulation: ITD sensitivity was measured in a two-alternative forced-choice discrimination task, using seven multi-electrode conditions. We hypothesized that by introducing low-rate ITDs at a few electrodes alongside high-rate ITDs at remaining electrodes provides sufficient ITD cues for sound localization. The present data suggests the binaural system of BiCI listeners can extract pertinent cues to achieve ITD sensitivity even when high-rate ITD information is presented at the majority of the cochlear locations. This lends to a possibility in implementation of ITD cues to current processing strategies.

Sound Source Identification Based on Functional Generalized Inverse Beamforming

Sound Source Identification Based on Functional Generalized Inverse Beamforming