Acoustic Background Noise Research Articles

Characterization of bearing health in a noisy environment using vibroacoustic approach

The bearing fault detection localized at outer race on a rotating machine operating in the same environment with another machine (whose wind tunnel generates a different noise) is investigated using acoustic and vibration analysis. The impact of increased acoustic and vibratory background noise on outer race defect detection is studied through spectral kurtogram representation and envelope analysis. The spectral power densities of these two types of noise clearly show that the nature of their associated radiation (airborne or solid-borne) has a different impact. The presence of a strongly amplified background noise demonstrates that the acoustic modality can detect the outer race defect peak at higher levels than the vibratory modality.

Comparison of quantitative [11C]PE2I brain PET studies between an integrated PET/MR and a stand-alone PET system

PET/MR systems demanded great efforts for accurate attenuation correction (AC) but differences in technology, geometry and hardware attenuation may also affect quantitative results. Dedicated PET systems using transmission-based AC are regarded as the gold standard for quantitative brain PET. The study aim was to investigate the agreement between quantitative PET outcomes from a PET/MR scanner against a stand-alone PET system.Nine patients with Parkinsonism underwent two 80-min dynamic PET scans with the dopamine transporter ligand [11C]PE2I. Images were reconstructed with resolution-matched settings using 68Ge-transmission (stand-alone PET), and zero-echo-time MR (PET/MR) scans for AC. Non-displaceable binding potential (BPND) and relative delivery (R1) were evaluated using volumes of interest and voxel-wise analysis.Correlations between systems were high (r ≥ 0.85) for both quantitative outcome parameters in all brain regions. Striatal BPND was significantly lower on PET/MR than on stand-alone PET (-7%). R1 was significantly overestimated in posterior cortical regions (9%) and underestimated in striatal (-9%) and limbic areas (-6%). The voxel-wise evaluation revealed that the MR-safe headphones caused a negative bias in both parametric BPND and R1 images. Additionally, a significant positive bias of R1 was found in the auditory cortex, most likely due to the acoustic background noise during MR imaging.The relative bias of the quantitative [11C]PE2I PET data acquired from a SIGNA PET/MR system was in the same order as the expected test–retest reproducibility of [11C]PE2I BPND and R1, compared to a stand-alone ECAT PET scanner. MR headphones and background noise are potential sources of error in functional PET/MR studies.

The dependence of transformer sound power measurement accuracy on microphone configurations in the anechoic chamber.

The paper is aimed at investigating the number and location of microphones required to accurately calculate the sound power level of a harmonic noise source. The measurement procedure performed is based on the IEC standard for the determination of transformer sound power level. The current procedure defines numerous measurement points around the tested transformer. Reducing the number of measurement points could significantly speed up the measurement procedure when a transformer is tested during the manufacturing process. Within the research work, estimations on the number and location of measurement points are identified to accurately calculate sound power level. The approach is tested and validated using specially developed software in LabView, that allows inclusion or exclusion of measurement points from the sound power level calculations. To prevent harmonic wave interference, acoustic wave reflections and background noise, the measurement procedure took place in the anechoic chamber.

In Vivo Measurements of the Bulk Ultrasonic Attenuation Coefficient of Breast Tissue Using a Novel Phase-Insensitive Receiver.

This study describes the first in vivo acoustic attenuation measurements of breast tissue undertaken using a novel phase-insensitive detection technique employing a differential pyroelectric sensor. The operation of the sensor is thermal in nature, with its output signal being dictated by the acoustic power integrated over its surface. The particularly novel feature of the sensor lies in its differential principle of operation, which significantly enhances its immunity to background acoustic and vibration noise. A large area variant of the sensor was used to detect ultrasonic energy generated by an array of 14 discrete 3.2-MHz plane piston transducers, transmitted through pendent breasts in water. The transduction and reception capability represent key parts of a prototype Quantitative Ultrasound Computed Tomography Test Facility developed at the National Physical Laboratory to study the efficacy of phase-insensitive ultrasound computed tomography of breast phantoms containing a range of appropriate inclusions, in particular, the measurement uncertainties associated with quantitative reconstructions of the acoustic attenuation coefficient. For this study, attenuation coefficient measurements were made using 1-D projections on 12 nominally healthy study volunteers, whose age ranged from 19 to 65 years. Averaged or bulk attenuation coefficient values were generated in the range 1.7-4.6 dBcm ^-1 at 3.2 MHz and have been compared with existing literature, derived from in vivo and ex vivo studies. Results are encouraging and indicate that the relatively simple technique could be applied as a robust method for assessing the properties of breast tissue, particularly the balance of fatty (adipose) and fibroglandular components.

Addressing federal court facility acoustical design challenges

Modern federal courthouse acoustical designs mandate a multidisciplinary approach in which architects, engineers, consultants, contractors, and court administrators work together to provide cost-effective designs which meet explicit speech privacy, room acoustic, and building systems background noise requirements. It is a primary responsibility of the acoustical consultant to understand and support implementation of owner standards, the U.S. Courts Design Guide criteria, and various tenant adjacency requirements, all of which present unique challenges. Owner standards vary from project to project in accordance with functional requirements, building location, economic constraints, and other project-specific needs. A new edition of the U.S. Courts Design Guide containing significant updates is expected to be published in the near future. Tenant adjacency requirements are often ambiguous and conflict with each other and with other project criteria. This paper examines a case study of three federal courthouse designs, focusing on challenges associated with intertwined project requirements, cost constraints, and the design-build process. The study translates federal judiciary and courthouse tenant requirements and provides insight into established design precedents. Solutions to design challenges are presented, which ensure future designs meet the acoustical needs of federal court facilities.

The low acoustic noise and turbulence wind tunnel of the University of Sao Paulo

AbstractThis paper introduces the Low Acoustic Noise and Turbulence (LANT) wind tunnel of the Sao Carlos School of Engineering, University of Sao Paulo (USP-EESC), Brazil. The closed-loop wind tunnel features several devices to improve flow uniformity, reduce swirl, and lower the background acoustic noise and turbulence, enabling stability and aeroacoustic experiments. The design criteria was based on the best practices reported, in particular for low turbulence wind tunnels. Yet, these criteria are conflicting and we discuss the decisions that had to be made and present flow quality results that were achieved. The 16-bladed axial fan with 13-blade stators is driven by a variable-speed electric motor. At the corners, 100 mm dense acoustic foam is installed on the vertical walls, floor and ceiling, and the turning vanes are filled with acoustic-absorbing material. The long settling chamber contains a 3.175 mm mesh hexagonal honeycomb and five fine mesh nylon screens, ending in a 7:1 area ratio short contraction. The 3-m long closed-working section has a $1\times 1\ {\rm m}^2$ cross-section area. At 15 m/s the working section wall boundary layer is less than 100 mm thick, providing an area of at least $800\times 800\ \mathrm{mm}^2$ where the streamwise flow uniformity was within 1%. In the 10–30 m/s flow speed range, the turbulence intensity ranged from 0.05% to 0.071% and the background acoustic noise level, obtained with an inflow microphone, ranged from 90 and 110 dB. A benchmark experiment on a flat plate boundary layer produced an almost perfect two-dimensional Blasius profile up to $Re_x \approx 2.5 \times 10^6$ . A beamforming benchmark experiment on aeroacoustics accurately identified the sound emitted by a cylinder immersed in the flow.

Advanced design of Close-Proximity (CPX) trailer enclosure acoustics on tyre/road noise measurement

The PolyU Mark II Twin-wheeled CPX trailer was developed for the measurement of tyre/road noise in Hong Kong urban environment according to a standard methodology (ISO/CD 11819-2) - the Close-Proximity (CPX) method. Numerical simulations of the acoustics of PolyU Mark II CPX enclosure were conducted and a good agreement between numerical and experimental results was obtained. In order to extend the capacity of the Mark II CPX trailer and enhance the acoustic performance within the enclosure for future tyre/road noise studies, the validated numerical simulations were carried on to design the next generation of the PolyU CPX system. Through analyzing the acoustic performance within the enclosures of different dimensions and the distributions of sound pressure level (SPL) inside the anechoic chamber, the geometry of the PolyU Mark III CPX enclosure was finally determined. With newly designed enhanced interior wall absorption, the new PolyU Mark III CPX enclosure design was delivered into numerical simulations for acoustic analysis. Fewer room modes and high uniformity of SPL distributions were observed within the new enclosure design. The PolyU Mark III CPX enclosure was fabricated based on the corresponding dimensions and the specific absorption layers. Great consistency was achieved between the numerical and measured results of the Mark III CPX enclosure. In addition, the PolyU Mark III CPX enclosure shows an improved acoustic property with a lower background noise level during road tests than Mark II CPX enclosure. The outcome of this study firmly establishes the feasibility of designing advanced CPX enclosure with numerical simulations with results that can be realized in realistic CPX measurement.

Listening Enhancement in Noisy Environments: Solutions in Time and Frequency Domain

The intelligibility of speech from a telephone or a public address system is often affected by acoustical background noise in the near-end listening environment. Speech intelligibility and listening effort can be improved by adaptive pre-processing of the loudspeaker signal. This is called Near-End Listening Enhancement (NELE). The speech spectrum is dynamically modified, taking the acoustical background noise at the near-end into account. In this paper, two opposite NELE strategies with either Noise-Masking-Proportional Shaping or Noise-Masking-Inverse Shaping are proposed which are appropriate for different noise characteristics. Both strategies are formulated in closed form in the frequency domain. They do not require to optimize an intelligibility measure but use explicitly the masking threshold. Motivated by the frequency domain approach, a simpler time domain solution is derived which is based on linear prediction techniques and does not need the masking calculations. The proposed NELE solutions outperform state-of-the-art in terms of computational complexity, memory requirement, continuous processor load, and latency.

Irrigation Control through Acoustic Proximal Sensing of the Onset of Surface Water

Irrigation is a useful crop enhancement procedure up to the point where free surface water appears. However, over-irrigation can lead to an accumulation of free water on the soil surface, which in turn results in overland flow and a high risk of contaminant loss. The current work addresses the problem of measuring free water on the surface of agricultural soils by a real-time acoustic remote sensing method. Directional acoustic transmitter and receiver arrays are used to define a “footprint” on the ground from which changes in reflectance are sensed. These arrays are mounted on a moving irrigator. Chirp signals are used to provide along-path resolution and to ensure robustness against unwanted acoustic background noise from farm machinery and the irrigator. Field measurements have been conducted above a well-defined “quadrat” with controlled and measured water content and also with the instrument mounted on an operational irrigator. A structured light camera mounted above the footprint is used to validate surface water fraction. It is found that the areal fraction of free water on the soil surface can be reliably estimated from changes in the amplitude of the reflected sound waves. The mechanism giving rise to the observed acoustic reflectivity changes is discussed and a model is developed which agrees with normalized intensity observations with a coefficient of determination R2 between 0.65 and 0.83. The rms error between model predictions and observations is comparable to the rms variation of the measurements, indicating that there is insignificant error due to the choice of model.

Multi-TALK: Multi-Microphone Cross-Tower Network for Jointly Suppressing Acoustic Echo and Background Noise.

In this paper, we propose a multi-channel cross-tower with attention mechanisms in latent domain network (Multi-TALK) that suppresses both the acoustic echo and background noise. The proposed approach consists of the cross-tower network, a parallel encoder with an auxiliary encoder, and a decoder. For the multi-channel processing, a parallel encoder is used to extract latent features of each microphone, and the latent features including the spatial information are compressed by a 1D convolution operation. In addition, the latent features of the far-end are extracted by the auxiliary encoder, and they are effectively provided to the cross-tower network by using the attention mechanism. The cross tower network iteratively estimates the latent features of acoustic echo and background noise in each tower. To improve the performance at each iteration, the outputs of each tower are transmitted as the input for the next iteration of the neighboring tower. Before passing through the decoder, to estimate the near-end speech, attention mechanisms are further applied to remove the estimated acoustic echo and background noise from the compressed mixture to prevent speech distortion by over-suppression. Compared to the conventional algorithms, the proposed algorithm effectively suppresses the acoustic echo and background noise and significantly lowers the speech distortion.

Digital enhancement of speech perception in noisy environments

If speech is reproduced in an acoustically disturbed environment via a loudspeaker, the perception in terms of intelligibility or listening effort may be impaired. Typical situations arise in connection with mobile phones or public address systems. The location of the listener is called the near-end while the speech signal is received from the far-end. Usually, the acoustical background noise cannot be influenced, but speech perception may be enhanced by digital pre-processing of the loudspeaker signal. This pre-processing technique is called Near-End Listening Enhancement (NELE). The received signal is filtered adaptively, by taking the instantaneous characteristics of the background noise into account and by exploiting psychoacoustic properties of the auditory system such as the masking threshold. The increase of the loudspeaker volume is often not allowed or very limited, to avoid injury of hearing or damages of the loudspeaker and because of the listening comfort. In this contribution, state-of-the-art NELE algorithms operating in the short-term frequency domain are reviewed and novel time domain approaches are presented. Furthermore, the differing NELE constraints of public address systems and mobile phones are discussed.

Auditory Enhancement of Illusory Contour Perception.

Illusory contours (ICs) are borders that are perceived in the absence of contrast gradients. Until recently, IC processes were considered exclusively visual in nature and presumed to be unaffected by information from other senses. Electrophysiological data in humans indicates that sounds can enhance IC processes. Despite cross-modal enhancement being observed at the neurophysiological level, to date there is no evidence of direct amplification of behavioural performance in IC processing by sounds. We addressed this knowledge gap. Healthy adults ( n = 15) discriminated instances when inducers were arranged to form an IC from instances when no IC was formed (NC). Inducers were low-constrast and masked, and there was continuous background acoustic noise throughout a block of trials. On half of the trials, i.e., independently of IC vs NC, a 1000-Hz tone was presented synchronously with the inducer stimuli. Sound presence improved the accuracy of indicating when an IC was presented, but had no impact on performance with NC stimuli (significant IC presence/absence× Sound presence/absence interaction). There was no evidence that this was due to general alerting or to a speed-accuracy trade-off (no main effect of sound presence on accuracy rates and no comparable significant interaction on reaction times). Moreover, sound presence increased sensitivity and reduced bias on the IC vs NC discrimination task. These results demonstrate that multisensory processes augment mid-level visual functions, exemplified by IC processes. Aside from the impact on neurobiological and computational models of vision, our findings may prove clinically beneficial for low-vision or sight-restored patients.

Bee Swarm Activity Acoustic Classification for an IoT-Based Farm Service.

Beekeeping is one of the widespread and traditional fields in agriculture, where Internet of Things (IoT)-based solutions and machine learning approaches can ease and improve beehive management significantly. A particularly important activity is bee swarming. A beehive monitoring system can be applied for digital farming to alert the user via a service about the beginning of swarming, which requires a response. An IoT-based bee activity acoustic classification system is proposed in this paper. The audio data needed for acoustic training was collected from the Open Source Beehives Project. The input audio signal was converted into feature vectors, using the Mel-Frequency Cepstral Coefficients (with cepstral mean normalization) and Linear Predictive Coding. The influence of the acoustic background noise and denoising procedure was evaluated in an additional step. Different Hidden Markov Models’ and Gaussian Mixture Models’ topologies were developed for acoustic modeling, with the objective being to determine the most suitable one for the proposed IoT-based solution. The evaluation was carried out with a separate test set, in order to successfully classify sound between the normal and swarming conditions in a beehive. The evaluation results showed that good acoustic classification performance can be achieved with the proposed system.

Preliminary analysis of operating conditions of acoustic sounder (SODAR) before the installation

The SOnic Detection And Ranging (SODAR) is a surface-based remote sensing instrument, based on Doppler effect and sound signal. This paper discusses the operating conditions of Monostatic SODAR system, error in the measured Atmospheric Boundary Layer (ABL) height by taking account of the antenna characteristics which are pulse transmission, receiving durations, spectral augmentation of the received echo and atmospheric refraction effects. The Monostatic SODAR is able to receive even very weak echo by providing a good directional response with high conversion efficiency. The location and orientation of the antenna are finalized after a detailed analysis of the site with relative to several parameters like comparative measurement of acoustic background noise and frequency response of noise. Also, before installation of the system, the transducer has to be individually characterized for it’s transmitted and received conversion efficiency, and variations in the product of these efficiencies (conversion gain). The antenna is systematically characterized, in an acoustic anechoic and reverberation chamber, with respect to its conversion efficiencies and directional response.

Time\u2013Frequency Feature Fusion for Noise Robust Audio Event Classification

This paper explores the use of three different two-dimensional time–frequency features for audio event classification with deep neural network back-end classifiers. The evaluations use spectrogram, cochleogram and constant-Q transform-based images for classification of 50 classes of audio events in varying levels of acoustic background noise, revealing interesting performance patterns with respect to noise level, feature image type and classifier. Evidence is obtained that two well-performing features, the spectrogram and cochleogram, make use of information that is potentially complementary in the input features. Feature fusion is thus explored for each pair of features, as well as for all tested features. Results indicate that a fusion of spectrogram and cochleogram information is particularly beneficial, yielding an impressive 50-class accuracy of over 96% in 0 dB SNR and exceeding 99% accuracy in 10 dB SNR and above. Meanwhile, the cochleogram image feature is found to perform well in extreme noise cases of -,5 dB and -,10 dB SNR.

Acoustic crypsis in southern right whale mother-calf pairs: infrequent, low-output calls to avoid predation?

Southern right whales (Eubalaena australis) invest substantial amounts of energy in their calves, while facing the risk of having them predated upon by eavesdropping killer whales (Orcinus orca). We tested the hypothesis that southern right whale mother-calf pairs employ acoustic crypsis to reduce acoustic detectability by such predators. Specifically, we deployed multi-sensor DTAGs on nine lactating whales for a total of 62.9 h in a Western Australian breeding ground, and used a SoundTrap to estimate the concomitant acoustic background noise. Vocalisations were recorded at low rates of <10 calls h-1 (1 call per dive) and at low received levels between 123±8 and 134±10 dB re. 1 µPa RMS depending on call type. We conclude that such acoustic crypsis in southern right whales and other baleen whales decreases the risk of alerting potential predators and hence jeopardizing a substantial energetic investment by the mother.

Convolutional Neural Networks to Enhance Coded Speech

Enhancing coded speech suffering from far-end acoustic background noise, quantization noise, and potentially transmission errors, is a challenging task. In this work we propose two postprocessing approaches applying convolutional neural networks (CNNs) either in the time domain or the cepstral domain to enhance the coded speech without any modification of the codecs. The time domain approach follows an end-to-end fashion, while the cepstral domain approach uses analysis-synthesis with cepstral domain features. The proposed postprocessors in both domains are evaluated for various narrowband and wideband speech codecs in a wide range of conditions. The proposed postprocessor improves speech quality (PESQ) by up to 0.25 MOS-LQO points for G.711, 0.30 points for G.726, 0.82 points for G.722, and 0.26 points for adaptive multirate wideband codec (AMR-WB). In a subjective CCR listening test, the proposed postprocessor on G.711-coded speech exceeds the speech quality of an ITU-T-standardized postfilter by 0.36 CMOS points, and obtains a clear preference of 1.77 CMOS points compared to legacy G.711, even better than uncoded speech with statistical significance. The source code for the cepstral domain approach to enhance G.711-coded speech is made available.

Low-Complexity Multi-Microphone Acoustic Echo Control in the Short-Time Fourier Transform Domain

Many modern communication and smart devices are equipped with several microphones, in addition to one or more loudspeakers. Each microphone not only acquires sounds produced in the near-end room, i.e., desired near-end speech, background noise, and other interferences, but also a far-end signal that is reproduced by the loudspeaker(s). This particular type of acoustic coupling, commonly denoted as acoustic echo, can be reduced in a distortionless manner by employing multi-microphone acoustic echo cancellation (MM-AEC) techniques. However, under noisy conditions, the performance of AEC is limited by the echo-to-noise ratio, and additional echo reduction is needed. Further, to ensure high-quality end-to-end communication in noisy environments, background noise has to be reduced as well. To achieve the latter, multi-microphone speech enhancement techniques, such as beamforming (BF), are often used as they are capable of reducing undesired signal components while causing little distortion to the desired near-end speech. In spite of its high computational cost, the most effective solution to reduce acoustic echoes and background noise is to cascade MM-AEC and BF. In this work, a low-complexity multi-microphone echo controller is introduced, which not only combines low-complexity MM-AEC with BF, but also integrates residual echo reduction into the beamformer design.

Smart Wristwatches Employing Finger-Conducted Voice Transmission System

The principal aim of this paper is to present a novel speech transmission system that conveys speech between an actuator in a wearable wristwatch and the ear bone of a user through a finger. If an individual wears a smart watch equipped with an actuator that can play speech sent via communication lines, speech vibrations propagate from the actuator to fingertips through the human tissue and bone. When an individual places his or her finger into their ear, speech conducted through the finger can be registered and heard. While listening to finger-conducted speech, sounds are muffled, significantly degrading the intelligibility of speech. To mitigate this problem, a formant enhancement filter is applied to the speech prior to being fed into the actuator. With this method, the impulse response of human tissue and bones between the fingertips and wrist on which the watch is worn is first estimated to account for speaker-dependent distortion. Based on the estimated impulse response, a gain filter is used to boost the sound spectra, especially within the formant regions, to compensate for frequency distortion prior to speech transmission. On the other hand, since the impulse responses of humans are quite different for each individual, we propose the novel idea of a personalized algorithm that guides users to select an appropriate gain filter, using the k-medoids clustering algorithm. Also, when an individual uses the proposed system, speech quality is degraded due to the ambient noise and acoustic echo between the microphone and actuator in the watch. Thus, to reduce background noise and acoustic echo, an integrated acoustic echo and background noise suppression algorithm is employed. Extensive simulations of the proposed system were performed by creating a novel phantom, which mimics the human hand with an aid of an ear simulator. We demonstrate that the proposed system has improved speech quality, when transmitting speech from the wearable wristwatch to a human perceptual organ through the finger.

Conversational speech levels and signal-to-noise ratios in realistic acoustic conditions.

Estimating the basic acoustic parameters of conversational speech in noisy real-world conditions has been an elusive task in hearing research. Nevertheless, these data are essential ingredients for speech intelligibility tests and fitting rules for hearing aids. Previous surveys did not provide clear methodology for their acoustic measurements and setups, were opaque about their samples, or did not control for distance between the talker and listener, even though people are known to adapt their distance in noisy conversations. In the present study, conversations were elicited between pairs of people by asking them to play a collaborative game that required them to communicate. While performing this task, the subjects listened to binaural recordings of different everyday scenes, which were presented to them at their original sound pressure level (SPL) via highly open headphones. Their voices were recorded separately using calibrated headset microphones. The subjects were seated inside an anechoic chamber at 1 and 0.5 m distances. Precise estimates of realistic speech levels and signal-to-noise ratios (SNRs) were obtained for the different acoustic scenes, at broadband and third octave levels. It is shown that with acoustic background noise at above approximately 69 dB SPL at 1 m distance, or 75 dB SPL at 0.5 m, the average SNR can become negative. It is shown through interpolation of the two conditions that if the conversation partners would have been allowed to optimize their positions by moving closer to each other, then positive SNRs should be only observed above 75 dB SPL. The implications of the results on speech tests and hearing aid fitting rules are discussed.