Virtual Sound Sources Research Articles

Mathematically trivial control of sound using a parametric beam focusing source

By exploiting a case regarded as trivial, this paper presents global active noise control using a parametric beam focusing source (PBFS). As with a dipole model, one is used for a primary sound source and the other for a control sound source, the control effect for minimizing a total acoustic power depends on the distance between the two. When the distance becomes zero, the total acoustic power becomes null, hence nothing less than a trivial case. Because of the constraints in practice, there exist difficulties in placing a control source close enough to a primary source. However, by projecting a sound beam of a parametric array loudspeaker onto the target sound source (primary source), a virtual sound source may be created on the target sound source, thereby enabling the collocation of the sources. In order to further ensure feasibility of the trivial case, a PBFS is then introduced in an effort to meet the size of the two sources. Reflected sound wave of the PBFS, which is tantamount to the virtual sound source output, aims to suppress the primary sound. Finally, a numerical analysis as well as an experiment is conducted, verifying the validity of the proposed methodology.

Method And Apparatus To Reproduce Wide Mono Sound

A wide mono sound reproducing method and apparatus to widen mono sound by using 2 channel speakers. The method include separating an input mono sound signal into a plurality of decorrelated signals, generating virtual sound sources by localizing each of the separated signals at virtual locations asymmetrical about a center of a front side of a listening point by applying different head related transfer functions to the separated signals, and canceling crosstalk of the generated virtual sound sources.

A Cinematic Spatial Sound Display for Panorama Video Applications

This paper describes a new sound spatialisation system which is an integral part of Rensselaer Polytechnic Institute’s new Experimental Media and Performing Arts Center (EMPAC). The Cinematic Spatial Sound Display (CSSD) was originally conceived for interactive panorama video installations, but the architecture goes beyond this particular application. The CSSD is characterised by its scalability to various loudspeaker configurations. It spatialises sound from dry sound files or live sources using control data that describe the spatial scenes. The time lines for source positions and other experimental parameters can be stored and edited in the CSSD, and the system can also process live user input to control selected parameters. The CSSD is more than just a sound positioning tool, and the underlying Virtual Microphone Control (ViMiC) technology was developed to support artists in designing new forms of spatial imagery. The software enables the user to create computer-generated rooms with virtual microphones and sound sources. The algorithm uses physical laws to auralise acoustic scenes – allowing realistic effects such as the Doppler shift and the simulation of various classical microphone techniques. Various parameters of ViMiC can be adjusted in real time, including the directivity patterns and orientations of both the microphones and sound sources as well as their precise locations. Surreal scenes can be created by assigning artificial directivity patterns to microphones or changing the laws of physics in the model. An algorithm to extract the sound-source positions in recordings using a microphone array is also part of the CSSD. The algorithm was specifically designed to operate in multiple sound-source scenarios and can also be used for telematic music applications.

Enhancing music with virtual sound sources

Enhancing music with virtual sound sources

The role of spectral components of the head-related transfer functions in evaluation of the virtual sound source motion in the vertical plane

Perception of virtual sound sources moving in the range of elevation from −45° to 45° (at zero azimuth) was studied with participation of listeners aged 57–73. The virtual sound source trajectory was created using nonindividualized head-related transfer functions and artificially synthesized spectral components specific to these functions. The percentage of correct responses in determining the direction of virtual motion depended on the way of imitation by increasing from low to high for the following succession of imitation methods: (i) displacement of the spectral minimum of broadband noise pulses within a frequency band of 5–12 kHz (the minimum retained a constant width of 1 kHz), (ii) variation of the spectral minimum width of noise pulses within 6–12 kHz, (iii) variation of the spectral maximum width within 4–8 kHz, (iv) simultaneous variation of the spectral minimum and maximum widths, (v) displacement of the spectral minimum and simultaneous variation of the spectral maximum width, and (vi) displacement of the spectral minimum and simultaneous variation of the spectral maximum width and power. For the latter stimulus, the mean percentage of correct responses (90 ± 5)% did not differ from the corresponding percentage (94 ± 3)% observed for the stimulus that was synthesized on the basis of nonindividualized head-related transfer functions and used as reference in synthesizing the spectral components.

Near-field characteristics of the parametric loudspeaker using ultrasonic transducers

A parametric speaker is a device for generating and focusing highly directional sound beams. It is essentially a by-product that comes with the nonlinearity of ultrasound. It is noteworthy that this directional beam was controlled and utilized mostly for far-field applications in the past. We empirically study the directivity and attenuation characteristics of the parametric loudspeaker in the near-field where we desire to use it. Physical parameters for experiments are imported from a theoretical model based on the far-field approximation. The findings are that increases in aperture size and modulation frequency cause higher directivity, but have more than twice the beamwidth of the far-field approximation. The attenuation also does not obey the inverse-square law which describes far-field spreading from acoustic sources. The results conclusively explain a series of formation and attenuation of the virtual sound sources and define limitations of use in the near-field.

Effects of individualized headphone equalization on front/back discrimination of virtual sound sources.

Individualized head related transfer functions (HRTFs) were used to process brief noise bursts for a two-interval forced choice front/back shift between virtual sound source locations presented via two models of headphones, the frequency responses of which could be made nearly flat for each of 21 listeners using their own individualized headphone equalization filters. In order to remove tone coloration differences between virtual sources processed using individualized HRTFs measured in front or in back of each listener, the spectral centroid of the “back” source was adjusted to more nearly match that of the source processed using the “front” HRTF. This manipulation resulted in chance levels of discrimination performance for 12 out of 21 listeners. For the remaining nine listeners showing good discrimination, the virtual sources presented using individualized headphone equalization supported significantly better front/back discrimination rates than did virtual sources presented without correction to headphone responses.

Implementation of sound ball with acoustic contrast control method.

It is well known that the problem of generating sound in the region of interest, by using finite number of speakers, is mathematically ill‐posed. This problem can be well‐posed; in other words, the way to drive the speakers to make desired sound field in the prescribed zone can be directly determined by using energy measure, which is called acoustic brightness, contrast control method. With the method, we can manuver sound ball or balls in space which can generate personal listening zone or virtual sound source. In this paper, a novel way that guides a way to design the speaker array in space to generate sound ball will be introduced. The signals between the zone of interest and the speakers are interpreted as vector spaces. The vector space interpretation certainly provides insight to design the arrays for acoustic brightness or contrast control. Theoretical formulation in vector space and experimental results of generating the sound ball will be introduced.

3-D localization of virtual sound sources: Effects of visual environment, pointing method, and training

The ability to localize sound sources in three-dimensional space was tested in humans. In Experiment 1, naive subjects listened to noises filtered with subject-specific head-related transfer functions. The tested conditions included the pointing method (head or manual pointing) and the visual environment (VE; darkness or virtual VE). The localization performance was not significantly different between the pointing methods. The virtual VE significantly improved the horizontal precision and reduced the number of front-back confusions. These results show the benefit of using a virtual VE in sound localization tasks. In Experiment 2, subjects were provided with sound localization training. Over the course of training, the performance improved for all subjects, with the largest improvements occurring during the first 400 trials. The improvements beyond the first 400 trials were smaller. After the training, there was still no significant effect of pointing method, showing that the choice of either head- or manual-pointing method plays a minor role in sound localization performance. The results of Experiment 2 reinforce the importance of perceptual training for at least 400 trials in sound localization studies.

Head related transfer functions for panned stereo audio content

A method to process audio signals, an apparatus accepting audio signals, a carrier medium that carried instructions for a processor to implement the method to process audio signals, and a carrier medium carrying filter data to implement a filter of audio signals. The method includes filtering a pair of audio input signals by a process that produces a pair of output signals corresponding to the results of: filtering each of the input signals with a HRTF filter pair, and adding the HRTF filtered signals. The HRTF filter pair is such that a listener listening to the pair of output signals through headphones experiences sounds from a pair of desired virtual speaker locations. Furthermore, the filtering is such that, in the case that the pair of audio input signals includes a panned signal component, the listener listening to the pair of output signals through headphones is provided with the sensation that the panned signal component emanates from a virtual sound source at a center location between the virtual speaker locations.

Effects of natural versus artificial spatial cues on electrophysiological correlates of auditory motion

The effect of the type of the auditory motion stimulus on neural correlates of motion processing were investigated using high-density electroencephalography. Sound motion was implemented by (a) gradual shifts in interaural time or (b) level difference; (c) motion of virtual 3D sound sources; or (d) successive activation of 45 loudspeakers along the horizontal plane. In a subset of trials, listeners ( N = 20) performed a two-alternative forced-choice motion discrimination task. Each trial began with a stationary phase of the acoustic stimulus in a central position, immediately followed by a motion of the stimulus. The motion onset elicited a specific cortical response that was dominated by large negative and positive deflections, the so-called change-N1 and change-P2. The temporal dynamics of these components depended on the auditory motion cues presented: Free-field motion and virtual 3D motion were associated with earlier cortical responses and with shorter reaction times than shifts in interaural time or level. Also, free-field motion elicited much stronger onset responses than simulated motion. These findings suggest that natural-like stimulation using stimuli presented in the free sound field allows more reliable conclusions on neural processing of sound motion, whereas artificial motion stimuli, in particular gradual shifts in interaural time or level, seem to be less suited with respect to this aim.

Comparison of Sound Pressure Distribution Determined by Numerical Analysis and Scaled-Up Experiment for Small Ultrasonic Probe with Lens

Ultrasonic medical equipment is used in not only diagnosis but also therapy such as that for treating hyperthermia. Many researchers are also studying high-frequency ultrasonic imaging systems with interluminal or catheter transducers. In both applications, an acoustic lens might improve the characteristics of ultrasonic medical probes. In this paper, a small acoustic lens for an ultrasonic catheter-type probe is described. The conventional three-dimensional finite-difference time-domain (3D FDTD) with orthogonal coordinates requires a large memory and a long calculation time to estimate the characteristics of the lens. To overcome these disadvantages, a simple two-dimensional (2D) FDTD calculation based on symmetry is proposed in this paper. A virtual spherical sound source whose amplitude distribution is equal to that of the sound propagation field of an actual sound source is also used to simplify the calculation. A numerical model of the lens with a lens holder is constructed. The experimental results agree well with the calculated results with the lens holder.

State-space models of head-related transfer functions for virtual auditory scene synthesis

This study investigates the use of reduced-order state-space models of collections of head-related transfer functions (HRTFs). Recent head-phone applications have motivated interest in binaural displays that can render multiple simultaneous virtual sound sources, acoustic reflections, and source and listener motion. In the present study, a multi-direction framework is considered that can render such phenomena by filtering source signals with a collection of HRTFs rather than individual HRTFs. The collection of HRTFs is implemented in the state-space, and approximation techniques are applied to construct low-order approximants that are indiscriminable from full-order HRTFs. Two experiments are described in which five observers are asked to discriminate between state-space and full-order renderings. Depending on the stimulus conditions and discrimination task, order thresholds of 7<or=N<or=24 are sufficient for an array of 50 HRTFs surrounding the listener. For comparison, a reference system of truncated minimum-phase impulse responses is also considered.

Head‐related transfer function enhancement for improved vertical‐polar localization.

Under ideal laboratory conditions, individualized head‐related transfer functions (HRTFs) can produce virtual sound localization performance approaching the level achieved with real sound sources in the free field. However, in real‐world applications of virtual audio, practical issues such as fit‐refit variability in the headphone response and nonindividualized HRTFs generally lead to much worse localization performance, particularly in the up‐down and front‐back dimensions. Here we present a new technique that “enhances” the localizability of a virtual sound source by increasing the spectral contrast of the acoustic features that are relevant for spatial perception within a set of locations with nearly identical binaural cues (i.e., a “cone‐of‐confusion”). Validation experiments show that this enhancement technique can improve localization accuracy across a broad range of conditions, with as much as a 33% reduction in vertical‐polar localization error for nonindividualized HRTFs measured on a KEMAR manikin; a 25% reduction in vertical‐polar error for nonindividualized HRTFs measured on other human listeners; and a 33% reduction in vertical‐polar error for individualized HRTFs presented under nonideal laboratory conditions (i.e., with headphone fit‐refit variability). These results suggest that the proposed technique could provide benefits across a wide range of real‐world virtual audio display applications. [Work sponsored by AFOSR.]

Effect of Emulated Head-Tracking for Reducing Localization Errors in Virtual Audio Simulation

Virtual audio simulation uses head-related transfer function (HRTF) synthesis and headphone playback to create a sound field similar to real-life environments. Localization performance is influenced by parameters such as the recording method and the spatial resolution of the HRTFs, equalization of the measurement chain as well as common headphone playback errors. The most important errors are in-the-head localization and front-back reversals. Among other cues, small movements of the head are considered to be important to avoid these phenomena. This study uses the BEACHTRON sound card and its HRTFs for emulating small head-movements by randomly moving the virtual sound source to emulate head-movements. This method does not need any additional equipment, sensors, or feedback. Fifty untrained subjects participated in the listening tests using different stimuli and presentation speed. A virtual target source was rendered in front of the listener by random movements of 1c-7deg. Experiments showed that this kind of simulation can be helpful to resolve in-the-head localization, but there is no clear benefit for resolving front-back errors. Emulation of small head-movements of 2deg could actually increase externalization rates in about 21% of the subjects while presentation speed is not significant.

Selective Listening Point Audio Based on Blind Signal Separation and Stereophonic Technology

SUMMARY A sound field reproduction method is proposed that uses blind source separation and a head-related transfer function. In the proposed system, multichannel acoustic signals captured at distant microphones are decomposed to a set of location/signal pairs of virtual sound sources based on frequency-domain independent component analysis. After estimating the locations and the signals of the virtual sources by convolving the controlled acoustic transfer functions with each signal, the spatial sound is constructed at the selected point. In experiments, a sound field made by six sound sources is captured using 48 distant microphones and decomposed into sets of virtual sound sources. Since subjective evaluation shows no significant difference between natural and reconstructed sound when six virtual sources and are used, the effectiveness of the decompos

Lateralization of virtual sound sources: Perception of starting and final points of motion trajectories

Lateralization of virtual sound sources: Perception of starting and final points of motion trajectories

3D-localization of virtual sound sources in normal-hearing and cochlear-implant listeners

The ability to localize sound sources in 3D-space was tested in humans. Five normal-hearing (NH) subjects listened via headphones to noises filtered with subject-specific head-related transfer functions. Four bilateral cochlear implant (CI) subjects listened via their clinical speech processors to noises filtered with subject-specific behind-the-ear head-related transfer functions. A virtual structured environment was presented via head mounted display. Two conditions were used: a condition where the subjects were naive and had no response feedback and a learning condition where the subjects were trained by providing extensive feedback during the test. Response feedback was provided via the visual virtual environment. The results show that the CI listeners performed generally worse than NH listeners, both in the horizontal and vertical dimensions. Both subject groups were able to learn to better localize sound sources, which is supported by lower localization errors in the learning condition. However, in the learning condition, the CI listeners showed a front/back confusion rate comparable to naive NH listeners, which was two times higher than for the trained NH listeners. These results indicate the necessity of new CI processing strategies which include spectral localization cues. Funding by FWF (P18401-B15).

Real-time auralization of modifiable rooms

Immersive virtual environments are a powerful tool for acousticians and architects to design buildings if the virtual reality system provides an interactive imaging of virtual sound sources with respect to the rooms' physical aspects. Current implementations using hybrid room acoustic simulation methods (e.g., combining image sources and ray tracing) enable the user to walk freely in such virtual architectural spaces, whereby the position/orientation of sound sources are interactively manipulable to detect possible acoustic defects, e.g., flutter echoes. In the case of coupled rooms, sound transmission effects must be included into the real-time simulation in order to identify deficient airborne sound insulation, whereby current implementations only support a change of state (open/closed) of fixed room-connecting elements, e.g., doors and windows. However, in scenarios like an architectural planning stage, it is convenient to manipulate the room geometry more freely, e.g., via the interactive positioning of stage reflector panels, but common spatial data structures, e.g., BSP- or Octrees do not efficiently support these operations. For this purpose, the concept of Spatial Hashing, which originates from computer graphics for collision detection of deformable objects, is applied to the simulation process. This adaptation also features an efficient identification and update process of image sources.

Measurement‐based fuzzy interpolation of room impulse responses

Application of room impulse responses (RIRs) to acoustic evaluation and auralization often requires many measurements to get enough information about the hall, or to provide enough flexibility for virtual sound source placements in convolution reverberation. In this paper we propose a measurement‐based fuzzy modeling method to approximate the RIR function at an arbitrary location between available measured points, without a priori information on the hall geometry or wall reflection parameters. For the fuzzy model identification we define an accuracy indicator of the spatial density of the source positions and predict the required number of them in a selected hall. This indicator quantifies the relationship of the early reflections, determined for various measured positions. This paper also proposes a method that treats nonuniform spatial sampling of the measurement positions, and its implementation for 2D cases is shown. Nonuniform spatial sampling can be useful when RIRs at some source positions ‐‐ e.g. ...