Set Of Microphones Research Articles

The effect of fan-type inducer blade tip geometry on cavitation behavior and noise in an industrial centrifugal pump

Cavitation is one of the biggest problems in industrial centrifugal pumps because it causes serious vibration and noise. Therefore, there is a demand for pump manufacturers to prevent problems due to the cavitation before delivering products to the site. Noise is considered to increase due to the development of cavitation at the tip of the inducer blades. In this study, the location of cavitation development was controlled by changing the inducer blade tip geometry to reduce noise. A centrifugal pump with a fan-type inducer was operated to measure noise at various cavitation numbers and flow rates with a microphone set up on the outer wall near the inducer. A visualization experiment using an acrylic casing around the inducer, a high-speed camera, and strobe light was also conducted to observe the behavior of cavitation development. The relationship between cavitation form and noise was investigated in detail.

A small-scale active anechoic chamber

This paper presents a solution method to suppress reflections from the walls in an anechoic room using active control, in which the control coefficients are obtained with an efficient algorithm. One of the main challenges in suppressing the reflections, is the estimation of the reflected sound field, which is a non-measurable quantity. The reflected sound field is computed using the Kirchhoff-Helmholtz integral with a set of microphones on a circle. The effectiveness of the method is shown on an experimental setup. The experimental setup is a small-scale version of an acoustic anechoic chamber. An anechoic chamber, with larger dimensions and higher frequencies, will result in a system with a relatively large number of sources and sensors. It is computationally expensive to find a set of control filters for larger-scale systems. The conjugate gradient algorithm using block-circulant preconditioning is an efficient method to solve for the optimal set of control filters. The preconditioned conjugate gradient algorithm is used together with the Kirchhoff-Helmholtz integral. Real-time measurements show that the reflections are effectively suppressed, expressed in terms of the reverberation time. This results in a small-scale active anechoic chamber.

Effectiveness of Television Streaming on Hearing Aid User Performance and Satisfaction in Quiet and in Noise.

The purpose of this study was to determine the effectiveness of television (TV) streaming on hearing aid user performance and satisfaction in quiet and in noise. Twenty experienced hearing aid users were evaluated in quiet and in noise (5 dB SNR) under three hearing aid conditions: hearing aids, streaming with hearing aid microphones attenuated by 6 dB, and streaming with hearing aid microphones muted. The audiovisual Connected Speech Test served as the signal, and the ipsilateral competing message from the Synthetic Sentence Identification-Ipsilateral Competing Message (SSI-ICM) served as the noise. Measures of speech understanding, satisfaction with sound quality and clarity, and preference were obtained. Speech understanding was significantly better in quiet than in noise, and streaming improved performance in noise when the hearing aid microphones were muted. Sound quality satisfaction was significantly better in quiet than in noise but was not impacted significantly by the hearing aid settings. Speech clarity satisfaction was significantly better in quiet than in noise, and streaming improved satisfaction in noise when the hearing aid microphones were muted. More participants preferred streaming with the microphones muted in quiet, in noise, and overall; however, results did not reach statistical significance. Hearing aid users did not benefit from TV streaming with the microphones attenuated or muted when listening in quiet. TV streaming was more beneficial to hearing aid users when listening in noise with the microphones muted. Hearing aid users should be counseled on the impact of the hearing aid microphone settings when using a TV streamer, particularly in noise.

Chladni Plate in Anechoic Chamber: Symmetry in Vibrational and Acoustic Response

A square metal plate (Chladni plate) is excited harmonically by a vibration shaker, while the whole system is set in an anechoic chamber to stop reflections, isolate the system from sound entering from the surroundings, and deal with direct sounds only. As far as the authors are aware, such an arrangement has not been achieved so far. Vibration modes are visualized by using poppy grains scattered over the upper surface of the plate and are also recorded by a camera located above it, inserted among the acoustic wedges on the roof of the chamber, which made it possible to record the patterns and avoid unpleasant sounds associated with some of them. Four distinctive vibration modes of the plate are then originally identified using vibrational and acoustic mode identification. These responses from the plate are measured both by an accelerometer attached to the central point of the plate and by a microphone set on the same vertical line as the accelerometer but above it, measuring the direct sound. The signals from the accelerometer and the microphone are then compared in two experimental arrangements, and their forms and the frequency contents are found to be equivalent. It is shown that the existing symmetry, i.e., the exact correspondence between vibrational and acoustic responses, can be used as the identifier of the patterns formed on the plate and the associated modal frequency.

Time-of-flight estimation in acoustic pyrometry: sensitivity to pulse characteristics

Acoustic pyrometry is a non-intrusive measurement technique that may have several applications in turbomachinery. This methodology estimates the gas temperature by measuring the time of flight of an acoustic wave moving through a medium. It can be accomplished by placing a sound source (emitter) and a set of microphones (receivers) on opposite sides of a section. The emitter generates a sound pulse, and the receivers detect it. Since the emitter-receiver distances are known and fixed, the average temperatures of the paths traversed by the acoustic pulse can be computed by estimating the time-of-flight through deconvolution techniques. However, despite the straightforward principle, an acoustic wave suffers a variation of amplitude when propagating within a medium because of energy losses and ambient noise. Hence, time-of-flight estimation becomes a critical task, especially when considering high-frequency waves or short distances between sensors. It is then fundamental to select proper acoustic waves to maximise the cross-correlation between the signals of the emitter-receiver couples, thus improving the accuracy of the time-of-flight measurements and, consequently, the estimation of the spatial temperature distribution within a specific area. This study is a preliminary investigation, based on a modelling approach, to estimate the impact of different acoustic waves on the accuracy of the time-of-flight measurement. The results of this analysis will be useful to design and setup an acoustic pyrometry application.

Urban Traffic Noise Analysis Using UAV-Based Array of Microphones.

(1) Background: Transition to smart cities involves many actions in different fields of activity, such as economy, environment, energy, government, education, living and health, safety and security, and mobility. Environment and mobility are very important in terms of ensuring a good living in urban areas. Considering such arguments, this paper proposes monitoring and mapping of a 3D traffic-generated urban noise emissions using a simple, UAV-based, and low-cost solution. (2) Methods: The collection of relevant sound recordings is performed via a UAV-borne set of microphones, designed in a specific array configuration. Post-measurement data processing is performed to filter unwanted sound and vibrations produced by the UAV rotors. Collected noise information is location- and altitude-labeled to ensure a relevant 3D profile of data. (3) Results: Field measurements of sound levels in different directions and altitudes are presented in the paperwork. (4) Conclusions: The solution of employing UAV for environmental noise mapping results in being minimally invasive, low-cost, and effective in terms of rapidly producing environmental noise pollution maps for reports and future improvements in road infrastructure.

Analysing changes in physiological response to different soundscape scenarios.

Noise pollution is one of the key environmental stressors leading to multiple health impacts for people and communities. Exposure to high noise levels may also be responsible for sudden emotional and physiological changes in humans. The experiment examines the change in Galvanic Skin Response (GSR) signals during exposure to pre-recorded soundscape scenarios. These scenarios were recorded in New Delhi, India and London, UK, using a binaural microphone set. The listening experiment was conducted in a laboratory, where 24 healthy individuals without any hearing impairment or any psychological issues participated. A total of 30 soundscape scenarios were presented to each participant in a randomised order. The continuous decomposition analysis is conducted to decompose that data into tonic and phasic components. The phasic component of the signal is used for the analysis. It is observed that skin conductance response (SCR) increases with changes in the overall loudness of the signal. Psychoacoustic indicators are used to identify the relationship between different acoustical characters of noise and changes in SCR. The perception-based survey suggests that GSR response is highly dependent on the pleasantness level of the soundscape scenario.

A Stethoscope for Drones: Transformers-Based Methods for UAVs Acoustic Anomaly Detection

Unmanned Aerial Vehicles and the increasing variety of their applications are raising in popularity. The growing number of UAVs, emphasizes the significance of drones’ reliability and robustness. Thus, there is a need for an efficient self-observing sensing mechanism to detect real-time anomalies in drone behavior. Previous works suggested prediction models from control theory, yet, they are complex by nature and hard to implement, while Deep Learning solutions are of great utility. In this paper, we propose a real-time framework to detect anomalies in drones by analyzing the sound emitted from them. For this purpose, we construct a hybrid Deep Learning based Transformer and a Convolutional Neural Network inspired by the well-known VGG architecture. Our approach is examined over a dataset that is collected from a single microphone set located on a micro drone in real-time. Our approach achieves an F1-score of 88.4% in detecting anomalies and outperforms the VGG-16 architecture. Moreover, the framework presented in this paper reduces the number of parameters of the well-known VGG-16 from 138M, into a shrunk version with 3.6M parameters only. Additionally, our real-time approach, results in a smaller number of parameters in the neural network, and yet yields high accuracy in anomaly detection in drones with an average inference time of 0.2 seconds per second. Moreover, with an earphone that weighs less than 100 grams on top of the UAV, our method is shown to be beneficial, even in extreme conditions such as a micro-size dataset that is composed of three hours of flight recordings. The presented self-observing method can be implemented by simply adding a microphone to drones and transmitting the captured audio for analysis to the remote control or performing it onboard the drone using a dedicated microcontroller.

Acoustic Thrust Estimation on Turbofan Engines

An acoustic time of flight sensor is operated across a turbofan exhaust plume and is calibrated to provide nozzle thrust in a ground-test cell. Such a sensor could potentially be used to monitor any degradation of engine performance in an on-wing application. Validation of the technique was performed on an AE3007-A1E turbofan engine. Using a pneumatic horn and a set of microphones, the acoustic time of flight across the exhaust plume was estimated. A correlation-based approach was then used to relate the acoustic time of flight to the thrust produced by the engine. The thrust estimates resulting from this single-receiver approach matched load cell data within a root-mean-square (RMS) error of 1.1% full scale. The same approach was also used to estimate the thrust of a TFE731-2B engine. In that case, the estimated thrust matched load cell data to within an RMS error of 1.5% full scale. The technique is shown to significantly improve on a previous technique that produced errors of 5% RMS full scale or greater. Possible explanations for the comparative inaccuracy of the previous technique are analyzed.

Experimental acoustic modal analysis of a tenor saxophone.

This paper presents an application of the Experimental Acoustic Modal Analysis (AMA) on a vintage 10 M Conn tenor saxophone. This technique originates from the traditional Modal Analysis which is commonly adopted to determine the dynamic behaviour of solid structures. The methodology is based on the measurement of Frequency Response Functions (FRFs) as a ratio of the acoustic pressure, measured by means of a set of microphones positioned along the saxophone body, and the volume velocity produced by an acoustic source and measured at the instrument input. The microphones are housed on the stripped saxophone keys by means of tailored three-dimensional printed adapters. The AMA makes use of the FRFs to extract modal parameters, such as resonance frequencies, loss factors, and mode shapes. These parameters pertain to the cavity modes of the saxophone passive resonator. The analysis has been performed for three different notes (B ♭, B, and C) and for two registers of the instrument in order to show the potential of the technique. Moreover, the influence of the mouthpiece volume on the cavity modes has been shown. The information obtained by AMA can find useful application in the validation of analytical or numerical models of this kind of musical instruments.

Influence of directional microphones on listening effort in middle ear implant users

Besides speech intelligibility, subjective listening effort is an important outcome for the success of hearing devices and their signal processing. The aim of the present study was to determine subjective listening effort for speech in a noisy background in patients with the active middle ear implant Vibrant Soundbridge (VSB) for omnidirectional and directional microphone settings, with and without occlusion of the contralateral ear. A total of 15patients using aVSB were measured using the adaptive categorical listening effort scaling (ACALES) method in aring of loudspeakers placed in an anechoic room. Different background noises from different directions and simultaneously presented sentences from the Oldenburg sentence test were combined in four different realistic acoustic scenes. The directional microphone program reduced median subjective listening effort only numerically compared to the omnidirectional microphone program in acoustic scenarios with diffuse noise and with low signal-to-noise ratios; however, this difference failed to reach statistical significance. When occluding the ear contralateral to the VSB, all investigated listening effort categories were measured at significantly higher signal-to-noise ratios than with access to both ears. Due to missing statistical significance in reduction of listening effort, this study delivered no recommendation for or against usage of the directional microphone program; however, reduced listening effort was shown for binaural listening in comparison to monaural listening. Therefore, patients should be encouraged to always listen with both ears for best results.

Beamforming and Single-Microphone Noise Reduction: Effects on Signal-to-Noise Ratio and Speech Recognition of Bimodal Cochlear Implant Users.

We have investigated the effectiveness of three noise-reduction algorithms, namely an adaptive monaural beamformer (MB), a fixed binaural beamformer (BB), and a single-microphone stationary-noise reduction algorithm (SNRA) by assessing the speech reception threshold (SRT) in a group of 15 bimodal cochlear implant users. Speech was presented frontally towards the listener and background noise was established as a homogeneous field of long-term speech-spectrum-shaped (LTSS) noise or 8-talker babble. We pursued four research questions, namely: whether the benefits of beamforming on the SRT differ between LTSS noise and 8-talker babble; whether BB is more effective than MB; whether SNRA improves the SRT in LTSS noise; and whether the SRT benefits of MB and BB are comparable to their improvement of the signal-to-noise ratio (SNR). The results showed that MB and BB significantly improved SRTs by an average of 2.6 dB and 2.9 dB, respectively. These benefits did not statistically differ between noise types or between the two beamformers. By contrast, physical SNR improvements obtained with a manikin revealed substantially greater benefits of BB (6.6 dB) than MB (3.3 dB). SNRA did not significantly affect SRTs per se in omnidirectional microphone settings, nor in combination with MB and BB. We conclude that in the group of bimodal listeners tested, BB had no additional benefits on speech recognition over MB in homogeneous noise, despite the finding that BB had a substantial larger benefit on the SNR than MB. SNRA did not improve speech recognition.

Evaluating the benefit of hearing aids with motion-based beamformer adaptation in a real-world setup

Objective Conventional directional hearing aid microphone technology may obstruct listening intentions when the talker and listener walk side by side. The purpose of the current study was to evaluate hearing aids that use a motion sensor to address listening needs during walking. Design Each participant completed two walks in randomised order, one walk with each of two hearing aid programs: (1) conventional beamformer adaptation that activated an adaptive, multiband beamformer in loud environments and (2) motion-based beamformer adaptation that activated a pinna-mimicking microphone setting when walking was detected. Participants walked along a pre-defined track and completed tasks assessing speech understanding and environmental awareness. Study Sample Participants were 22 older adults with moderate-to-severe hearing loss and experience using hearing aids. Results More participants preferred the motion-based than conventional beamformer adaptation for speech understanding, environmental awareness, overall listening, and sound quality (p < 0.05). Measures of speech understanding (p < 0.01) and localisation of sound stimuli (p < 0.05) were significantly better with motion-based than conventional beamformer adaptation. Conclusions The results suggest that hearing aid users can benefit from beamforming that uses motion sensor input to adapt the signal processing according to the user’s activity. The real-world setup of this study had limitations.

The multi-functional rotor aerodynamic and aeroacoustic test platform at HKUST

This paper describes the multi-functional rotor noise and aerodynamics test platform at the Hong Kong University of Science and Technology (HKUST). To investigate the noise characteristics of propellers with aerodynamic flows, the test rig is installed in the 2.5×2 (m) low-speed and low-noise wind tunnel in the Aerodynamic and Acoustic Facility (AAF) at HKUST. The wind tunnel can facilitate flow from 0 to 40 m/s. The test rig is assembled in a turntable on the ceiling of the tunnel wall, which enables the testing range of pitch angle can vary from 0° (axial flow) to 90° (parallel flow), with an accuracy of 0.1°. The noise produced by the rotor is measured by a set of wall-mounted surface microphones. Semi-empirical calibration is conducted to quantify the noise reflection by the tunnel walls. A low-noise struct has been designed and manufactured to locate a set of far-field microphones equipped with nosecone, to improve the quality of acoustic measurement inside the flow. In addition, a synchronized system is developed to conduct the phase-locking Particle Image Velocimetry (PIV) measurement on the rotor, to study the flow pattern to better understand the noise generation mechanism.

Symmetrical and asymmetrical directional benefits are present for talkers at the front and side

Objective The purpose of the study was to examine the effects of symmetrical and asymmetrical directional microphone settings on speech recognition, localisation and microphone preference in listening conditions with on- and off-axis talkers. Design A within-subjects repeated-measure evaluation of three hearing aid microphone settings (bilateral omnidirectional, bilateral directional, asymmetrical directional) was completed in a moderately reverberant laboratory. An exploratory analysis of the potential relationship between microphone preference and unaided measures was also completed. Study Sample Twenty adult listeners with mild to moderately severe bilateral hearing loss participated. Results The directional and asymmetric microphone settings resulted in equivalent benefits for sentence recognition in noise, word recall, and localisation speed regardless of the speech loudspeaker location (on- or off-axis). However, localisation accuracy was significantly worse with the asymmetric fitting than the directional setting when speech was presented from the rear hemisphere. Listeners who always preferred directional microphones had significantly poorer unaided speech recognition than those who preferred the omnidirectional setting for one or more listening condition. Conclusions Benefits from directional and asymmetric processing were small in the current study, but generally similar to each other. Unaided speech recognition in noise performance may have utility as a clinical predictor of preference for directional processing.

Active control of the field scattered by the rigid wall of a semi-anechoic room—Simulations and full-scale off-line experiment

This paper is dedicated to an experimental implementation of active anechoicity control illustrating previous theoretical work. The aim of this experimental is to cancel the pressure scattered by the single rigid wall of a semi-anechoic room, turning the latter into a fully anechoic one over a given frequency range. The proposed method is presented in two steps: estimation of the scattered pressure and its control. Two numerical simulations are presented in order to show the feasibility of the method. An experimental campaign is presented, where the active control system leads to a reduction of the field scattered by the rigid wall throughout a significant part of the measuring volume inside the room. Although these results are based on post-processing of measurements, and not real-time computation, they confirm the possibility to extend the frequency range of a small facility toward lower frequencies, by adding to its passive lining a set of ordinary sources and pressure microphones.

New efficient subband structures for blind source separation

Convolutive mixtures of signals generated by more than one source and acquired by a set of microphones are commonly found in acoustic signal processing applications. Subband methods have been proposed to reduce computational complexity and improve the convergence rate of adaptive algorithms developed for blind source separation of these mixtures, without significantly impairing steady-state performance. Oversampled discrete Fourier transform (DFT) filter bank is a usual option for the generation of subband signals in order to avoid harmful aliasing effects, thereby maintaining sufficient samples to estimate the subband signal statistics. In this paper, two new subband structures, composed of cosine-modulated filter banks (CMFB) with critical or oversampled sampling and low-order adaptive subfilters, are proposed for efficient blind source separation approach in convolutive mixtures of speech signals. A time-varying step-size procedure that provides better convergence rate for several reverberation characteristics is advanced. In addition, both computational complexity and steady-state performance of the proposed structures are compared to those of the standard fullband algorithm, to other subband structures with oversampling and critical sampling, and to frequency domain algorithm. The advanced solutions are capable of improving the source interference ratio (SIR) by more than 5 dB. Finally, two strategies are presented to minimize the effects of remaining aliasing in the proposed subband approaches, which obtain an additional gain of about 3 dB in the asymptotic SIR.

The impact of cochlear implant microphone settings on the binaural hearing of experienced cochlear implant users with single-sided deafness

ObjectiveCochlear implantation has become a well-accepted treatment option for people with single-sided deafness (SSD) and has become a clinical standard in many countries. A cochlear implant (CI) is the only device which restores binaural hearing. The effect of microphone directionality (MD) settings has been investigated in other CI indication groups, but its impact on speech perception in noise has not been established in CI users with SSD. The focus of this investigation was, therefore, to assess binaural hearing effects using different MD settings in CI users with SSD.MethodsTwenty-nine experienced CI users with SSD were recruited to determine speech reception thresholds with varying target and noise sources to define binaural effects (head shadow, squelch, summation, and spatial release from masking), sound localization, and sound quality using the SSQ12 and HISQUI19 questionnaires. Outcome measures included the MD settings “natural”, “adaptive”, and “omnidirectional”.ResultsThe 29 participants involved in the study were divided into two groups: 11 SONNET users and 18 OPUS 2/RONDO users. In both groups, a significant head shadow effect of 7.4–9.2 dB was achieved with the CI. The MD setting “adaptive” provided a significant head shadow effect of 9.2 dB, a squelch effect of 0.9 dB, and spatial release from masking of 7.6 dB in the SONNET group. No significant summation effect could be determined in either group with CI. Outcomes with the omnidirectional setting were not significantly different between groups. For both groups, localization improved significantly when the CI was activated and was best when the omnidirectional setting was used. The groups’ sound quality scores did not significantly differ.ConclusionsAdaptive directional microphone settings improve speech perception and binaural hearing abilities in CI users with SSD. Binaural effect measures are valuable to quantify the benefit of CI use, especially in this indication group.

Effects of Directionality, Compression, and Working Memory on Speech Recognition.

Previous research has shown that the association between hearing aid-processed speech recognition and individual working memory ability becomes stronger in more challenging conditions (e.g., higher background noise levels) and with stronger hearing aid processing (e.g., fast-acting wide dynamic range compression, WDRC). To date, studies have assumed omnidirectional microphone settings and collocated speech and noise conditions to study such relationships. Such conditions fail to recognize that most hearing aids are fit with directional processing that may improve the signal to noise ratio (SNR) and speech recognition in spatially separated speech and noise conditions. Here, we considered the possibility that directional processing may reduce the signal distortion arising from fast-acting WDRC and in turn influence the relationship between working memory ability and speech recognition with WDRC processing. The combined effects of hearing aid processing (WDRC and directionality) and SNR were quantified using a signal modification metric (cepstral correlation), which measures temporal envelope changes in the processed signal with respect to a linearly amplified reference. It was hypothesized that there will be a weaker association between working memory ability and speech recognition for hearing aid processing conditions that result in overall less signal modification (i.e., fewer changes to the processed envelope). Twenty-three individuals with bilateral, mild to moderately severe sensorineural hearing loss participated in the study. Participants were fit with a commercially available hearing aid, and signal processing was varied in two dimensions: (1) Directionality (omnidirectional [OMNI] versus fixed-directional [DIR]), and (2) WDRC speed (fast-acting [FAST] versus slow-acting [SLOW]). Sentence recognition in spatially separated multi-talker babble was measured across a range of SNRs: 0 dB, 5 dB, 10 dB, and quiet. Cumulative signal modification was measured with individualized hearing aid settings, for all experimental conditions. A linear mixed-effects model was used to determine the relationship between speech recognition, working memory ability, and cumulative signal modification. Signal modification results showed a complex relationship between directionality and WDRC speed, which varied by SNR. At 0 and 5 dB SNRs, signal modification was lower for SLOW than FAST regardless of directionality. However, at 10 dB SNR and in the DIR listening condition, there was no signal modification difference between FAST and SLOW. Consistent with previous studies, the association of speech recognition in noise with working memory ability depended on the level of signal modification. Contrary to the hypothesis above, however, there was a significant association of speech recognition with working memory only at lower levels of signal modification, and speech recognition increased at a faster rate for individuals with better working memory as signal modification decreased with DIR and SLOW. This research suggests that working memory ability remains a significant predictor of speech recognition when WDRC and directionality are applied. Our findings revealed that directional processing can reduce the detrimental effect of fast-acting WDRC on speech cues at higher SNRs, which affects speech recognition ability. Contrary to some previous research, this study showed that individuals with better working memory ability benefitted more from a decrease in signal modification than individuals with poorer working memory ability.

Estimating Real-World Performance of Percutaneously Coupled Bone-Conduction Device Users With Severe-to-Profound Unilateral Hearing Loss.

Purpose The bone-conduction device attached to a percutaneous screw (BCD) is an important treatment option for individuals with severe-to-profound unilateral hearing loss (UHL). Clinicians may use subjective questionnaires and speech-in-noise measures to evaluate BCD use in this patient population; however, the translation of these metrics to real-world aided performance is unclear. The purpose of this study was twofold: first, to measure speech-in-noise performance in BCD users with severe-to-profound UHL in a simulated real-world environment, relative to individuals with normal hearing bilaterally; second, to determine if BCD users' subjective reports of aided performance relate to simulated real-world performance. Method A between-subjects design with two groups was conducted with 14 adults with severe-to-profound UHL (BCD group) and 10 age-matched participants with normal hearing bilaterally (control group). Speech-in-noise tests were administered in an eight-speaker R-Space simulating a real-world environment. To further explore speech-in-noise evaluation methods for this population, testing was also completed in a clinically common two-speaker array. The effects of various microphone settings on performance were explored for BCD users. Subjective performance was measured with the Abbreviated Profile of Hearing Aid Benefit (APHAB; Cox & Alexander, 1995) and the Speech, Spatial and Qualities of Hearing Scale (Gatehouse & Noble, 2004). Statistical analyses to explore relationships between variables included repeated-measures analysis of variance, regression analyses, independent-samples t tests, nonparametric Mann-Whitney tests, and correlations. Results In the simulated real-world environment, BCD group participants struggled with speech-in-noise understanding compared to control group participants. BCD benefit was observed for all microphone settings when speech stimuli were presented to the side with the BCD. When adaptive directional or fixed directional microphone settings were used, a relationship was noted between simulated real-world speech-in-noise performance for speech stimuli presented to the side with the BCD and subjective reports on the Background Noise subscale of the APHAB. Conclusions The Background Noise subscale of the APHAB may help estimate real-world speech-in-noise performance for BCD users with severe-to-profound UHL for signals of interest presented to the implanted side, specifically when adaptive or fixed directional microphone settings are used. This subscale may provide an efficient and accessible alternative to assessing real-world speech-in-noise performance in lieu of less clinically available measurement tools, such as an R-Space.