Position Of Sound Source Research Articles

Binaural Localization of Multiple Sound Sources by Non-Negative Tensor Factorization

This paper presents non-negative factorization of audio signals for the binaural localization of multiple sound sources within realistic and unknown sound environments. Non-negative tensor factorization (NTF) provides a sparse representation of multichannel audio signals in time, frequency, and space that can be exploited in computational audio scene analysis and robot audition for the separation and localization of sound sources. In the proposed formulation, each sound source is represented by means of spectral dictionaries, temporal activation, and its distribution within each channel (here, left and right ears). This distribution, being dependent on the frequency, can be interpreted as an explicit estimation of the Head-Related Transfer Function (HRTF) of a binaural head which can then be converted into the estimated sound source position. Moreover, the semisupervised formulation of the non-negative factorization allows us to integrate prior knowledge about some sound sources of interest whose dictionaries can be learned in advance, whereas the remaining sources are considered as background sound, which remains unknown and is estimated on the fly. The proposed NTF-based sound source localization is applied here to binaural sound source localization of multiple speakers within realistic sound environments.

Localization of marine seismic vibrator based on hyperbolic Radon transform

In marine seismic surveys to explore seafloor resources, the structure below the seafloor is estimated from the obtained sound waves, which are emitted by a marine seismic sound source and reflected or refracted between the layers below the seafloor. In order to estimate the structure below the seafloor from returned waves, information of the sound source position and the sound speed are needed. Marine seismic vibrators, which are one of the marine seismic sound sources, have some advantages such as high controllability of the frequency and phase of the sound, and oscillation at a high depth. However, when the sound source position is far from the sea surface, it becomes difficult to specify the exact position. In this paper, we propose a method to estimate the position of a marine seismic vibrator and the sound speed from obtained seismic data by formulating an optimization problem via hyperbolic Radon transform. Numerical simulations confirmed that the proposed method almost achieves theoretical lower bounds for the variances of the estimations.

A long term effect on speech recognition in the patients with simultaneous bilateral cochlear implants

Objective: To investigate an effect on speech recognition after bilateral cochlear implants(CI) simultaneously. Methods: Nine subjects who underwent bilateral CI operation simultaneously in Peking Union Hospital in 2007 were assigned as bilateral group, another 9 subjects with unilateral CI were chosen as unilateral group according to the age, gender, duration of deafness. Hearing threshold, speech recognition of phrases, disyllabic words, single word in quiet and noise environment were calculated, respectively. Three different sound source positions were set up to simulate the three effects of binaural hearing (head shadow, binaural redundancy and binaural squelch) in noise environment. The speech signal intensity was 70 dBSPL, the signal to noise ratio (SNR) was 0 dB, + 4 dB, + 8 dB, and the speech recognition of phrases were examined in bilateral CI group. All the data was analyzed by SPSS 19.0 software. Results: In quiet environment, the average aid-hearing threshold was significantly reduced as (7.2±3.0)dB in bilateral CI group compared to unilateral CI group(P<0.05). However, the speech recognition of phrases, disyllabic words, and single word in quiet and noise environment did not show significant differences between bilateral CI and unilateral CI groups (P>0.05). In the testing under noise environment, the speech recognition of bilateral CI group was significantly higher than that of unilateral CI group (P<0.05). In the estimation of noise environment (SNR=+ 8), the phrase test and speech recognition of head shadow model in bilateral groups were significantly better than that of unilateral group (P<0.01). Further, bilateral CI group showed statistical significances in binaural redundancy and binaural squelch models(P<0.01), there was statistically significant were also found difference between two groups (P<0.01); Bilateral CI group did not reveal significant difference compared with unbilateral CI group in binaural squelch model(P>0.05). Under different signal-to-noise ratio (SNR=+ 8, + 4) noise environment, all the three given models in bilateral CI group show higher signal-to-noise ratio, higher speech recognition rate (Independent-Samples t test, P<0.01). In addition, head shadow model presented negative influences on speech recognition rate compared with binaural redundancy and binaural squelch model by SNR(P<0.05). SNR influences on binaural redundancy and binaural squelch model were almost identical, F=0.371, P=0.549. Conclusion: The subjects who underwent simultaneous bilateral cochlear implantation could significantly reduce aid-hearing threshold, and greatly improve the speech recognition of phrases, disyllabic words, and single word under noise environment.

Sonic Interactions in Virtual Reality: State of the Art, Current Challenges, and Future Directions

A high-fidelity but efficient sound simulation is an essential element of any VR experience. Many of the techniques used in virtual acoustics are graphical rendering techniques suitably modified to account for sound generation and propagation. In recent years, several advances in hardware and software technologies have been facilitating the development of immersive interactive sound-rendering experiences. In this article, we present a review of the state of the art of such simulations, with a focus on the different elements that, combined, provide a complete interactive sonic experience. This includes physics-based simulation of sound effects and their propagation in space together with binaural rendering to simulate the position of sound sources. We present how these different elements of the sound design pipeline have been addressed in the literature, trying to find the trade-off between accuracy and plausibility. Recent applications and current challenges are also presented.

Effect of mesoscale eddies on deep-water sound propagation

The mesoscale eddies are the most common mesoscale phenomena in the ocean. The cold and warm water carried by the mesoscale eddies can significantly change the structure of the sound speed field, which is the key factor affecting long range deep-water sound propagation. In this presentation, a deep-sea analytical eddy model [Henrick et al., JASA 62, 860–870(1977) ] is used to determine the sound speed distributions produced by cold eddy in the southwest of the South China Sea. The statistical characteristics of the influence of eddies on the acoustic propagation are investigated with a range-dependent parabolic equation (PE) model to asses the effects of the relative position of sound sources and eddies, the frequency sensitivity, the acoustic energy deviation of the convergence zone and the shadow zone for long-range (600 km) low frequency (50 Hz) sound propagation in this ocean model. Specific features of the induced sound propagation variability such as the travel time deviation and the standard deviation of ensemble intensity Will be quantified. Finally, the refraction due to mesoscale eddies on large scale low-frequency propagation is discussed using the full three-dimensional parabolic (FOR3D) and explaining by the full three-dimensional ray theory (Bellhop3D).

Research of Space Positioning Method Based on Sound Field HBT Interference

Based on Hanbury Brown-Twiss (HBT) interference in the sound field, a space positioning method is presented to realize the long-distance and high-precision positioning of sound sources in media. Firstly, theoretical model of HBT interference positioning is established. Location of the sound source can be acquired by analyzing the correlation function of the output signals. Then, sound source localization under different signal-to-noise ratios (SNR) shows that by this method, the sound source can be accurately found with six sensors (two arrays) even the SNR is low to 0.04. Positioning experiment in air is carried out, and the experimental results show that the sound source can be accurately located at 42 meters, and the positioning error is low to 0.1 meters. Thus the validity and accuracy of the HBT interference space location principle is demonstrated. It provides new ideas for the research of long-range target location in sound propagation media (air, water, etc.).

Proposal of sound source position estimation method by portable sound direction estimation device

Proposal of sound source position estimation method by portable sound direction estimation device

Localization Experiments with Reporting by Head Orientation: Statistical Framework and Case Study

This research focuses on sound localization experiments in which subjects report the position of an active sound source by turning toward it. A statistical framework for the analysis of the data is presented together with a case study from a large-scale listening experiment. The statistical framework is based on a model that is robust to the presence of front/back confusions and random errors. Closed-form natural estimators are derived, and one-sample and two-sample statistical tests are described. The framework is used to analyze the data of an auralized experiment undertaken by nearly nine hundred subjects. The objective was to explore localization performance in the horizontal plane in an informal setting and with little training, which are conditions that are similar to those typically encountered in consumer applications of binaural audio. Results show that responses had a rightward bias and that speech was harder to localize than percussion sounds, which are results consistent with the literature. Results also show that it was harder to localize sound in a simulated room with a high ceiling despite having a higher direct-to-reverberant ratio than other simulated rooms.

Applications of operator theory and symmetry arguments to sound field recording and reproduction

The problem of recording and reproducing sound fields using microphone/loudspeaker arrays is treated by exploiting its symmetries. This approach leads to well-known results, such as high-order Ambisonics corresponding to the requirement of the error field to be invariant under rotations. More importantly, it allows us to treat some difficult problems critical to the application of sound field recording and reproduction to virtual reality, where one or more users can freely move around a realistic environment. In this presentation, we consider three of these problems. First, we present a method to extrapolate extended sound field information from conventional spherical microphone array recordings by imposing symmetry constrains based on a priori knowledge of the sound source positions. Second, we apply the same symmetry constrains to the problem of sound field reproduction, allowing us to formulate a method of presenting spatial sound to moving listeners. Finally, we consider the problem of describing and reproducing the sound field due to a moving sound source.

Cepstral distance based channel selection for distant speech recognition

Shifting from a single to a multi-microphone setting, distant speech recognition can be benefited from the multiple instances of the same utterance in many ways. An effective approach, especially when microphones are not organized in an array fashion, is given by channel selection (CS), which assumes that for each utterance there is at least one channel that can improve the recognition results when compared to the decoding of the remaining channels. In order to identify this most favourable channel, a possible approach is to estimate the degree of distortion that characterizes each microphone signal. In a reverberant environment, this distortion can vary significantly across microphones, for instance due to the orientation of the speaker’s head. In this work, we investigate on the application of cepstral distance as a distortion measure that turns out to be closely related to properties of the room acoustics, such as reverberation time and direct-to-reverberant ratio. From this measure, a blind CS method is derived, which relies on a reference computed by averaging log magnitude spectra of all the microphone signals. Another aim of our study is to propose a novel methodology to analyze CS under a wide set of experimental conditions and setup variations, which depend on the sound source position, its orientation, and the microphone network configuration. Based on the use of prior information, we introduce an informed technique to predict CS performance. Experimental results show both the effectiveness of the proposed blind CS method and the value of the aforementioned analysis methodology. The experiments were conducted using different sets of real and simulated data, the latter ones derived from synthetic and from measured impulse responses. It is demonstrated that the proposed blind CS method is well related to the oracle selection of the best recognized channel. Moreover, our method outperforms a state-of-the-art one, especially on real data.

Representation of Multidimensional Stimuli: Quantifying the Most Informative Stimulus Dimension from Neural Responses.

A common way to assess the function of sensory neurons is to measure the number of spikes produced by individual neurons while systematically varying a given dimension of the stimulus. Such measured tuning curves can then be used to quantify the accuracy of the neural representation of the stimulus dimension under study, which can in turn be related to behavioral performance. However, tuning curves often change shape when other dimensions of the stimulus are varied, reflecting the simultaneous sensitivity of neurons to multiple stimulus features. Here we illustrate how one-dimensional information analyses are misleading in this context, and propose a framework derived from Fisher information that allows the quantification of information carried by neurons in multidimensional stimulus spaces. We use this method to probe the representation of sound localization in auditory neurons of chinchillas and guinea pigs of both sexes, and show how heterogeneous tuning properties contribute to a representation of sound source position that is robust to changes in sound level.SIGNIFICANCE STATEMENT Sensory neurons' responses are typically modulated simultaneously by numerous stimulus properties, which can result in an overestimation of neural acuity with existing one-dimensional neural information transmission measures. To overcome this limitation, we develop new, compact expressions of Fisher information-derived measures that bound the robust encoding of separate stimulus dimensions in the context of multidimensional stimuli. We apply this method to the problem of the representation of sound source location in the face of changes in sound source level by neurons of the auditory midbrain.

Stereophonic Microphone Array for the Recording of the Direct Sound Field in a Reverberant Environment

State-of-the-art stereo recording techniques using two microphones have two main disadvantages: first, a limited reduction of the reverberation in the direct sound component, and second, compression or expansion of the angular position of sound sources. To address these disadvantages, the aim of this study is the development of a true stereo recording microphone array that aims to record the direct and reverberant sound field separately. This array can be used within the recording and playback configuration developed in Grosse and van de Par, 2015. Instead of using only two microphones, the proposed method combines two logarithmically-spaced microphone arrays, whose directivity patterns are optimized with a superdirective beamforming algorithm. The optimization allows us to have a better control of the overall beam pattern and of interchannel level differences. A comparison between the newly-proposed system and existing microphone techniques shows a lower percentage of the recorded reverberance within the sound field.

Numerical simulation and experimental research of spectral noise characteristics for the four-wheel landing gear

Numerical simulation and wind tunnel experiment were conducted for flow field and flow noise characteristics of four-wheel landing gear. Under the landing speed, large eddy simulation method was applied to simulate the unsteady turbulent flow field around the landing gear. In addition, vortex-acoustic theory was also employed to compute the intensity and position of sound source, and integral method of FW-H equation was used to solve the sound field generated from different components and their combinations, analyze the noise generation mechanism, spectral characteristics and far-field directivity. Meanwhile, the contribution of each component to the total noise was assessed. The aerodynamic acoustic experiment was conducted for landing gear in the wind tunnel, and a microphone was applied to measure and obtain the spectral characteristics of noise, which was compared with the numerical computation result. As shown from the result, a good agreement was found between the experimental and numerical results. Besides, it was shown from the simulation result that small parts ignored in the computational sound field played a greater impact on prediction result of noise, especially high-frequency noise. As shown from landing gear vorticity and sound pressure distribution, sound source was mainly distributed in the solid surface of various components for the landing gear. Additionally, sound source was also occurred in the vortex shedding positions behind each component. Greater energy and density of the sound source were found in the gap formed between tire and bogie and in the pillar stairs. The tire noise showed maximum contribution to the total noise, followed by the bogie noise, while the pillar noise was minimal. The total noise spectrum was closest to the tire noise spectrum.

The Reduction of Vertical Interchannel Crosstalk: The Analysis of Localisation Thresholds for Natural Sound Sources

In subjective listening tests, natural sound sources were presented to subjects as vertically-oriented phantom images from two layers of loudspeakers, ‘height’ and ‘main’. Subjects were required to reduce the amplitude of the height layer until the position of the resultant sound source matched that of the same source presented from the main layer only (the localisation threshold). Delays of 0, 1 and 10 ms were applied to the height layer with respect to the main, with vertical stereophonic and quadraphonic conditions being tested. The results of the study showed that the localisation thresholds obtained were not significantly affected by sound source or presentation method. Instead, the only variable whose effect was significant was interchannel time difference (ICTD). For ICTD of 0 ms, the median threshold was −9.5 dB, which was significantly lower than the −7 dB found for both 1 and 10 ms. The results of the study have implications both for the recording of sound sources for three-dimensional (3D) audio reproduction formats and also for the rendering of 3D images.

Effect of Discretization of Sound Source Position on Sense of Presence in 3D Audio Systems Using Multiple Vertical Panning( Hearing and Virtual Reality)

Effect of Discretization of Sound Source Position on Sense of Presence in 3D Audio Systems Using Multiple Vertical Panning( Hearing and Virtual Reality)

Effect of preceding stimulation on sound localization and its representation in the auditory midbrain

Prior stimulation can influence the perception of sound source location. Some psychophysical sound localization procedures differ in the amount of prior stimulation, which may affect measures of localization accuracy. If and how particularly the number of preceding stimuli affects sound localization and the neural representation of sound source position has not been investigated so far and will be the focus of the present report. We trained Mongolian gerbils in a left/right discrimination task where the target stimulus was preceded by silence or followed a number of reference stimuli. Localization thresholds decreased with the number of references presented before the target stimulus. The smallest thresholds were found after the presentation of a train of 5 reference stimuli and after silence. We recorded from units in the inferior colliculus (IC) of anaesthetized gerbils using virtual-acoustic space stimuli mimicking the ones used in the behavioural task and applied signal detection theory to compare behavioural and neurometric thresholds. We found that neurometric thresholds based on spike rate information of single units covered a wide range of threshold values but only neurometric thresholds that were based on responses of small populations of IC units reached consistently thresholds we also observed in the behavioural experiment. Unlike behavioural thresholds, however, neurometric thresholds were independent of the number of reference stimuli suggesting that processing stages downstream from the IC might better reflect the effect of prior stimulation.

Azimuth Tracking of Underwater Moving Sound Source Based on Time Delay Estimation Using Hydrophone Array

Hydrophones array represents underwater sensors that can be used to detect azimuth of target by obtain the time delay information. Success rate of time delay estimation is influenced by characteristics of sound propagation like reflection, scattering, and reverberation. Simulation and experiment of hydrophone array in real environment aimed to analyze performance of the cross correlation (CC) and generalized cross correlation with the phase transform (PHAT) weighting to estimate time delay between two signals. The objective of the simulation is to know the influence of time delay against various SNR by CC and PHAT algorithm. The algorithm then tested in a pond to detect stationary and moving position of sound source. Result of the simulation and experiment in real environment which is considered as scatter condition showed that PHAT performed better than CC. In moving sound experiments, tracking the azimuth of the mini vessel can be done by time delay estimation using PHAT.

Distance-varying filters to synthesize head-related transfer functions in the horizontal plane from circular boundary values

This paper derives a continuous-space model to describe variations in magnitude of complex head-related transfer functions (HRTFs) along angles and radial distances throughout the horizontal plane. The radial part of this model defines a set of horizontal-plane distance-varying filters (HP-DVFs) that are used to synthesize the HRTFs for arbitrary sound source positions on the horizontal plane from initial HRTFs obtained for positions on a circular boundary at a single distance from the head of a listener. The HP-DVFs are formulated in terms of horizontal-plane solutions to the three-dimensional acoustic wave equation, which are derived by assuming invariance along elevation angles in spherical coordinates. This prevents the free-field inaccurate distance decay observed when assuming invariance along height in cylindrical coordinates. Furthermore, discontinuities along the axis connecting the ears are also overcome, which appear when assuming invariance along the polar angle in interaural coordinates. This paper also presents a magnitude-dependent band-limiting threshold (MBT) for restricting the action of filters to a limited angular bandwidth, which is necessary in practice to enable discrete-space models that consider a finite number of sources distributed on the initial circle. Numerical experiments using a model of a human head show that the overall synthesis accuracy achieved with the proposed MBT outperforms the one achieved with the existing frequency-dependent threshold, especially at low frequencies and close distances to the head.

A Psychoacoustic Investigation on the Effect of External Shading Devices on Building Facades

Due to energetic and natural lighting factors, building facades often present external shading devices, but the acoustic properties of such devices have not yet been well studied. This study was carried out using a full-scale model of a portion of a shading device, in a semi-anechoic chamber, using traditional and sound absorbing louvres. The psychoacoustic effects produced by the shading system were evaluated through comparisons between averaged values of loudness, roughness and sharpness levels, as well as sound pressure levels as reference. Results highlighted that the sound absorbing shading device offers good attenuation in terms of loudness, roughness and sound pressure level, with a small reduction in sharpness. The traditional shading system studied does not efficiently reduce the analysed parameters, or even worsens the situation. Several analyses of variance were carried out, one for each situation studied. The sound source position and the louvres’ tilt angle both produce statistically significant effects on almost all of the variations of the parameters studied. The analyses of the partial eta squared factors highlighted that source position and louvre tilt angle affect the variations of the parameters studied to a different degree in respect of the two types of louvres.

Sound source localization method in an environment with flow based on Amiet–IMACS

A sound source localization method is proposed to localize and analyze the sound source in an environment with airflow. It combines the improved mapping of acoustic correlated sources (IMACS) method and Amiet's method, and is called Amiet–IMACS. It can localize uncorrelated and correlated sound sources with airflow. To implement this approach, Amiet's method is used to correct the sound propagation path in 3D, which improves the accuracy of the array manifold matrix and decreases the position error of the localized source. Then, the mapping of acoustic correlated sources (MACS) method, which is as a high-resolution sound source localization algorithm, is improved by self-adjusting the constraint parameter at each irritation process to increase convergence speed. A sound source localization experiment using a pair of loud speakers in an anechoic wind tunnel under different flow speeds is conducted. The experiment exhibits the advantage of Amiet–IMACS in localizing a more accurate sound source position compared with implementing IMACS alone in an environment with flow. Moreover, the aerodynamic noise produced by a NASA EPPLER 862 STRUT airfoil model in airflow with a velocity of 80m/s is localized using the proposed method, which further proves its effectiveness in a flow environment. Finally, the relationship between the source position of this airfoil model and its frequency, along with its generation mechanism, is determined and interpreted.