Higher Order Ambisonics Research Articles

Methodology for 3D sound synthesis of directional acoustic sources by higher-order ambisonics

This paper presents the 3D sound field synthesis using both the multimodal method and higher-order ambisonics (HOA) of the pressure field radiated by directional acoustic sources. Ambisonics is a technique for encoding and reproducing measured or modeled (virtual) sound pressure field, based on a decomposition of the acoustic field over spherical harmonics. The directional source considered in this work is an acoustic horn excited by a flat piston. The free-field radiation from this horn is first modeled accurately over a wide frequency range using the multimodal method, which requires relatively low computational resources. This radiated pressure field, collected on a dual-layer sphere of virtual sensors distributed over a Lebedev geometry, allows its projection into the ambisonic domain. The pressure field is then synthetized in the laboratory's 3D 5th order HOA spatialization sphere, which consists of fifty-six loudspeakers. This offers the ability of listening to the radiated sound using a higher-order ambisonic synthesis of a 'virtual' source before it is manufactured. To qualitatively evaluate the performance of the proposed procedure, the transfer function of the synthesized horn is measured around the listening point within the spatialization sphere. Key words : 3D sound synthesis, high-order ambisonics, directive sources, waveguides, horn, multimodal method

Using soundscape simulation to evaluate compositions for a public space sound installationa).

While urban sound management often focuses on sound as a nuisance, soundscape research suggests that proactive design approaches involving sound art installations can enhance public space experience. Nevertheless, there is no consensus on a methodology to inform the composition of sound installations through soundscape evaluation, and little research on the effect of composition strategies on soundscape evaluation. The present study is part of a research-creation collaboration around the design of a permanent sound installation in an urban public space in Paris (Niches Acoustiques by Nadine Schütz). We report on a laboratory study involving the evaluation of composition sketches prior to the deployment of the installation on-site. Participants familiar with the public space (N = 20) were exposed to Higher-Order Ambisonics recordings of the site, to which compositions of the sound installation pertaining to different composition strategies were added using a soundscape simulation tool. We found three principal components relevant for evaluating and comparing sound installation sketches: pleasantness, familiarity and variety. Further, all composition sketches had a significant effect on the soundscape's familiarity and variety, and the effect of the compositions on these two components was stronger when composition strategies involved abstract sounds (sounds which were not clearly identifiable).

Recording, Processing, and Reproduction of Vibrations Produced by Impact Noise Sources in Buildings

Several studies on the perception of impact sounds question the correlation of standardized approaches with perceived annoyance, while more recent studies have come to inconsistent conclusions. All these studies neglected the aspect of whole-body vibrations, which are known to be relevant for the perception of low-frequency sound and can be perceived especially in lightweight constructions. Basically, the contribution of vibrations to impact sound annoyance is still unknown and could be the reason for the contradictory results. To investigate this aspect, we measured vibrations on different types of floors under laboratory conditions and in situ. For this purpose, a vibration-sensing device was developed to record vibrations more cost-effectively and independently of commercial recording instruments. The vibrations of predefined impact sequences were recorded together with the sound field using a higher-order ambisonics microphone. In addition, a vibration exposure device was developed to expose the test objects to the exact vibrations that occur in the built environment. The vibration exposure device is integrated into the ambisonics reproduction system, which consists of a large number of loudspeakers in a spherical configuration. The article presents the development and performance achieved using the vibration-sensing unit and the vibration exposure device. The study is relevant for conducting future impact sound listening tests under laboratory conditions, which can be extended to include the reproduction of vibrations.

Sound Source Localization in Spherical Harmonics Domain Based on High-Order Ambisonics Signals Enhancement Neural Network

Sound Source Localization in Spherical Harmonics Domain Based on High-Order Ambisonics Signals Enhancement Neural Network

Modeling and reproduction of sound field by moving complex sound source

Previously, Wave Field Synthesis (WFS), Near-field Compensated Higher-Order Ambisonics (NFC-HOA), Boundary Surface Control (BoSC), and the Spectral Division Method (SDM) have been proposed as sound field synthesis methods. These methods are based on matching the acoustic physical quantities at the boundary of the reproduced sound field with those of the desired sound field. Sound field synthesis techniques can be classified into data-based and model-based methods. The former aim at reproducing sound fields captured by microphone arrays, while the latter aim at synthesizing sound fields based on mathematical models. Most existing sound field reproduction methods only consider simple sound source models, such as static point sources, instead of sound sources with complex radiation properties. To generate a more realistic sound image, this study investigated the formulation of a sound field generated by a sound source with complex radiation properties moving along arbitrary trajectories in a spherical harmonic domain, and reproduced the sound field using loudspeaker arrays.

Upmix B-Format Ambisonic Room Impulse Responses Using a Generative Model

Ambisonic room impulse responses (ARIRs) are recorded to capture the spatial acoustic characteristics of specific rooms, with widespread applications in virtual and augmented reality. While the first-order Ambisonics (FOA) microphone array is commonly employed for three-dimensional (3D) room acoustics recording due to its easy accessibility, higher spatial resolution necessitates using higher-order Ambisonics (HOA) in applications such as binaural rendering and sound field reconstruction. This paper introduces a novel approach, leveraging generative models to upmix ARIRs. The evaluation results validate the model’s effectiveness at upmixing first-order ARIRs to higher-order representations, surpassing the aliasing frequency limitations. Furthermore, the spectral errors observed in the Binaural Room Transfer Functions (BRTFs) indicate the potential benefits of using upmixed ARIRs for binaural rendering, significantly improving rendering accuracy.

Computational aspects of real-time auralizations of jazz venues

Historically, architectural acoustics has focused on concert halls for European Classical music, even though other genres, including rock, blues, and Jazz, have large groups of followers as well. Additionally, the acoustic requirements for European Classical music differ fundamentally from Jazz and other music. For example, Jazz venues are typically smaller and dryer than a traditional European Classical concert hall. A recent survey showed that reverberation in these venues plays a secondary role to parameters that describe the affordance of a space to allow communication between improvising musicians. This presentation focuses on the unique computational auralization aspects of Jazz venues, including wave-based approaches for smaller enclosures, low-latency requirements, intelligent accompanying systems that can adapt ad-hoc to jazz soloists, and the use of anechoic jazz recordings as source material. The adequacies of different reproduction systems, from higher-order ambisonics and wave-field synthesis to headphone-based systems, will also be discussed. [Work supported by James West Fellowship, Leo and Gabriella Beranek Scholarship, and NSF HCC-1909229.]

A real-time auralization framework for scientific experiments

The software Virtual Acoustics (VA) is an open-source auralization framework developed at the Institute for Hearing Technology and Acoustics, RWTH Aachen University, Germany. Using physics-based models for source emission and sound propagation, it allows plausible rendering of virtual scenes based on purely synthetic data. Its modular design provides a variety of rendering modules using different assumptions for the sound propagation. This allows the rendering of various indoor and outdoor scenarios with adjustable degree of complexity. Based on a similarly modular concept, VA allows a flexible way of reproducing the sound via headphones or loudspeaker arrays. For this purpose, the signals can be spatialized using binaural synthesis, higher-order ambisonics or VBAP. The respective scenarios, including source and receiver movement, can be controlled using the C + +, C#, Matlab, Python programming languages. Additionally, there are interfaces for the Unreal and Unity game engines allowing the combination with sophisticated visual cues, e.g., using HMDs. The software’s flexible structure and configuration possibilities enable the controlled application in listening experiments while providing an adequate level of plausibility. This work discusses the options as well as the limitations of the software and presents application examples in scientific experiments.

Sound field synthesis for psychoacoustic research: In situ evaluation of auralized sound pressure levela).

The use of virtual acoustic environments has become a key element in psychoacoustic and audiologic research, as loudspeaker-based reproduction offers many advantages over headphones. However, sound field synthesis methods have mostly been evaluated numerically or perceptually in the center, yielding little insight into the achievable accuracy of the reproduced sound field over a wider reproduction area with loudspeakers in a physical, laboratory-standard reproduction setup. Deviations from the ideal free-field and point-source concepts, such as non-ideal frequency response, non-omnidirectional directivity, acoustic reflections, and diffraction on the necessary hardware, impact the generated sound field. We evaluate reproduction accuracy in a 61-loudspeaker setup, the Simulated Open Field Environment, installed in an anechoic chamber. A first measurement following the ISO 8253-2:2009 standard for free-field audiology shows that the required accuracy is reached with critical-band-wide noise. A second measurement characterizes the sound pressure reproduced with the higher-order Ambisonics basic decoder, with and without max rE weighting, vector base amplitude panning, and nearest loudspeaker mapping on a 187 cm × 187 cm reproduction area. We show that the sweet-spot size observed in measured sound fields follows the rule kr≤N/2 rather than kr≤N but is still large enough to avoid compromising psychoacoustic experiments.

A hearing aid “test drive”: Using virtual acoustics to accurately demonstrate hearing aid performance in realistic environments

A common challenge for audiologists when fitting hearing aids is that quiet clinics are unlike the noisy, reverberant places where patients report difficulty. This prevents direct user feedback about hearing aid performance. Virtual acoustics can allow a patient to experience a hearing aid in such environments, without leaving the clinic. A virtual reality (VR) audio-visual demonstration has been created which “test drives” hearing aid features in real-world scenarios, dynamically rendered over custom wired hearing aids and headphones. The scenes were created from 360° photographs and acoustic measurements made in real rooms. A room acoustic model was used for the simulation, tuned to match the measured data. Audio was rendered using principal component-base amplitude panning (PCBAP), which can provide dynamic VR audio with broadband magnitude and phase accuracy. This talk will provide comparisons between PCBAP and higher-order Ambisonics (HOA) for rendering hearing aid beamformers in anechoic and reverberant environments. Results show that PCBAP can render a beamformer’s directivity broadband with 2 to 3 dB error (95%) using only 36 filters in both anechoic and reverberant conditions. Even seventh-order HOA (64 filters) does not achieve similar performance. The talk will also include videos of the demonstration, highlighting viability for clinical applications.

Ambisonics decoder design based on sparse plane wave decomposition

Sound field reproduction by loudspeaker array is an important way to create acoustical virtual reality. High order ambisonics (HOA) is a highly applicable method to reproduce a desired sound field with a loudspeaker array. For a target sound field that is represented by a group of harmonic coefficients, the process of calculating loudspeaker weights for reproduction with these harmonic coefficients is called decoding. Mode-matching based decoding is usually applied in HOA reproduction and good performance is achieved in the loudspeaker array center (the ‘sweet spot’). However, the accuracy and sweet spot size are strongly affected by the recording order of the target sound field. To improve the reproduction performance, a new ambisonic decoding method is proposed based on the sparse plane wave decomposition in the case that the target sound field is steered by a small number of acoustic sources in this paper. The recorded target sound field is decomposed into a group of plane waves in the harmonic domain. Loudspeaker weights are calculated as a summation of the precalculated corresponding plane waves weights. Simulations are conducted in the free field and reverberation room to verify that the proposed method outperforms the mode-matching decoding. Experimental validation is conducted in a real listening room with a circular loudspeaker array. It is concluded that the reproduction error by the proposed method is superior to that by the sectorial mode matching for the investigated 2.5D reproduction case in the reverberant environment after 500 Hz and in terms of the MAC values, the proposed method ensures the spatial correspondence over frequency of the desired sound field and the reproduced sound field.

Audiovisual time-to-collision estimation for accelerating vehicles: The acoustic signature of electric vehicles impairs pedestrians' judgments

To avoid collisions, pedestrians intending to cross a road need to accurately estimate the time-to-collision (TTC) of an approaching vehicle. For TTC estimation, auditory information can be considered particularly relevant when the approaching vehicle accelerates. The sound of vehicles with internal combustion engine (ICEVs) provides characteristic auditory information about the acceleration state (increasing rotational speed and engine load). However, for electric vehicles (EVs), the acoustic signature during acceleration is less salient. Although the auditory detection of EVs has been studied extensively, there is no research on potential effects of the altered acoustic signature of EVs on TTC estimation. To close this gap, we compared TTC estimates for ICEVs and for EVs with and without activated acoustic vehicle alerting system (AVAS). We implemented a novel interactive audiovisual virtual-reality system for studying the human perception of approaching vehicles. Using acoustic recordings of real vehicles as source signals, the dynamic spatial sound field corresponding to a vehicle approaching in an urban setting is generated based on physical modeling of the sound propagation between vehicle and pedestrian (listener) and is presented via sound field synthesis (higher-order Ambisonics). In addition to the auditory simulations, the scene was visually presented on a head-mounted display with head tracking. Participants estimated the TTC of vehicles that either approached at a constant speed or accelerated positively. In conditions with constant speed, TTC estimates for EVs with and without AVAS were similar to those for ICEVs. In contrast, for accelerating vehicles, there was a substantial effect of the vehicle type on the TTC estimates. For the EVs, the mean TTC estimates showed a significant overestimation. Thus, subjects on average perceived the time of arrival of the EV at their position as longer than it actually was. The extent of overestimation increased with acceleration and presented TTC. This pattern is similar to a first-order TTC estimation representing a failure to consider the acceleration, which is consistently reported in the literature for visual-only presentations of accelerating objects. In comparison, the overestimation of TTC was largely reduced for the accelerating ICEVs. The AVAS somewhat improved the TTC estimates for the accelerating EVs, but without reaching the same level of accuracy as for the ICEVs. In real traffic scenarios, overestimations of the TTC of approaching vehicles might lead to risky road-crossing decisions. Therefore, our finding that pedestrians are significantly less able to use the acoustic information emitted by accelerating EVs for their TTC judgments, compared to accelerating ICEVs, has important implications for road safety and for the design of AVAS technologies.

The impact of head-related impulse response delay treatment strategy on psychoacoustic cue reconstruction errors from virtual loudspeaker arrays.

Known errors exist in loudspeaker array processing techniques, often degrading source localization and timbre. The goal of the present study was to use virtual loudspeaker arrays to investigate how treatment of the interaural time delay (ITD) cue from each loudspeaker impacts these errors. Virtual loudspeaker arrays rendered over headphones using head-related impulse responses (HRIRs) allow flexible control of array size. Here, three HRIR delay treatment strategies were evaluated using minimum-phase loudspeaker HRIRs: reapplying the original HRIR delays, applying the relative ITD to the contralateral ear, or separately applying the HRIR delays prior to virtual array processing. Seven array sizes were simulated, and panning techniques were used to estimate HRIRs from 3000 directions using higher-order Ambisonics, vector-base amplitude panning, and the closest loudspeaker technique. Compared to a traditional, physical array, the prior HRIR delay treatment strategy produced similar errors with a 95% reduction in the required array size. When compared to direct spherical harmonic (SH) fitting of head-related transfer functions (HRTFs), the prior delays strategy reduced errors in reconstruction accuracy of timbral and directional psychoacoustic cues. This result suggests that delay optimization can greatly reduce the number of virtual loudspeakers required for accurate rendering of acoustic scenes without SH-based HRTF representation.

On the selection of the number of beamformers in beamforming-based binaural reproduction

In recent years, spatial audio reproduction has been widely researched with many studies focusing on headphone-based spatial reproduction. A popular format for spatial audio is higher order Ambisonics (HOA), where a spherical microphone array is typically used to obtain the HOA signals. When a spherical array is not available, beamforming-based binaural reproduction (BFBR) can be used, where signals are captured with arrays of a general configuration. While shown to be useful, no comprehensive studies of BFBR have been presented and so its limitations and other design aspects are not well understood. This paper takes an initial step towards developing a theory for BFBR and develops guidelines for selecting the number of beamformers. In particular, the average directivity factor of the microphone array is proposed as a measure for supporting this selection. The effect of head-related transfer function (HRTF) order truncation that occurs when using too many beamformer directions is presented and studied. In addition, the relation between HOA-based binaural reproduction and BFBR is discussed through analysis based on a spherical array. A simulation study is then presented, based on both a spherical and a planar array, demonstrating the proposed guidelines. A listening test verifies the perceptual attributes of the methods presented in this study. These results can be used for more informed beamformer design for BFBR.

Spaced A-B placements of higher-order Ambisonics microphone arrays: Techniques for recording and balancing direct and ambient sound

Spaced A-B placements of higher-order Ambisonics microphone arrays: Techniques for recording and balancing direct and ambient sound

Sparse DNN Model for Frequency Expanding of Higher Order Ambisonics Encoding Process

The performance of higherorder Ambisonics (HOA) signals obtained using spherical harmonics decomposition method is disturbed by two primary sources of errors, the noise pollution in low-frequency band and the spatial aliasing in high-frequency band. Inspired by the HOA signals upscale method, which is performed using the sparse character of the sound field, this paper propose a sound field decomposition model based on a sparse deep neural network that offers HOA signals with wider frequency bandwidth. We use the frequency domain multi-scale convolutional network to realize the spherical harmonics decomposition, as well as learning the spatial aliasing pattern, based on which the aliasing-free HOA signals can be derived. Besides, we apply a sparse encoding network to cpature the sparse feature of the sound field which will improve the model performance when the sparse condition is satisfied. The experiments results prove that the proposed model can obtain HOA signals with wider frequency range of operation under multiple sources (up to 10 sources) and low reverberant environments (<inline-formula><tex-math notation="LaTeX">$T_{60}\le$</tex-math></inline-formula> 400 ms). When the sparsity feature cannot be satisfied (<inline-formula><tex-math notation="LaTeX">$T_{60} =$</tex-math></inline-formula> 800 ms), the proposed network model still maintain the same performance as the traditional methods.

Binaural-centered mode-matching method for enhanced reproduction accuracy at listener's both ears in sound field reproduction.

This paper proposes a binaural-centered mode-matching (BCMM) method that performs sound field reproduction with two reproduction points, one at each ear. A sound field reproduced by higher-order Ambisonics converges to the target field around a sweet spot whose size is inversely proportional to the frequency when the truncated order of the spherical harmonic expansion is constant. By contrast, BCMM translates the spherical harmonic coefficients on the basis of the addition theorem to realize two reproduction points at both ear positions. The BCMM method thereby avoids degradation of reproduction due to a smaller sweet spot at higher frequencies, as occurs with the conventional method, and thereby, leads to an accurate reproduction at higher frequencies. A comparison of numerical simulations of the binaural signals obtained when rendering with BCMM and conventional methods shows that, compared with the conventional method, the BCMM method improves the reproduction accuracy.

MPEG Standards for Compressed Representation of Immersive Audio

The term “immersive audio” is frequently used to describe an audio experience that provides the listener the sensation of being fully immersed or “present” in a sound scene. This can be achieved via different presentation modes, such as surround sound (several loudspeakers horizontally arranged around the listener), 3D audio (with loudspeakers at, above, and below listener ear level), and binaural audio to headphones. This article provides an overview of two recent standards that support the bitrate-efficient carriage of high-quality immersive sound. The first is MPEG-H 3D audio, which is a versatile standard that supports multiple immersive sound signal formats (channels, objects, and higher order ambisonics) and is now being adopted in broadcast and streaming applications. The second is MPEG-I immersive audio, an extension of 3D audio, currently under development, which is targeted for virtual and augmented reality applications. This will support rendering of fully user-interactive immersive sound for three degrees of user movement [three degrees of freedom (3DoF)], i.e., yaw, pitch, and roll head movement, and for six degrees of user movement [six degrees of freedom (6DoF)], i.e., 3DoF plus translational x, y, and z user position movements.

Spatial release from masking in varying spatial acoustic under higher order ambisonic-based sound reproduction system

A previous study found that spatial release from masking (SRM) could be observed under virtual reverberant environments using a first order Ambisonic-based sound reproduction system, however, poor localisation accuracy made it difficult to examine effect of varying reverberation time on SRM. The present study follows on using higher order Ambisonics (HOA) to examine how benefits from SRM vary in different spatial acoustics. Subjective speech intelligibility was measured where four room acoustics:reverberation time (RT)= 0.7 s (clarity (C50)= 16 dB, 7 dB); RT= 1.8 s (C50= 8 dB, 2 dB) were simulated via a third order Ambisonic system with a 16 channel spherical loudspeaker array. The masker was played from 8 azimuthal angles (0, +-45, +-90, +-135, 180 degrees) while the target speech was played from 0 degree. The listeners are deemed to benefit from SRM if their intelligibility scores were higher when the masker comes from a different angle than that of the target. We found while listeners could benefit from SRM at C50 = 16 dB and 8 dB, the benefit starts to diminish at C50 = 7 dB, and listeners could no longer benefit from SRM at C50 = 2 dB.

Sound Field Reproduction by Combination of Circular and Spherical Higher-Order Ambisonics: Part II—Hybrid System

Sound Field Reproduction by Combination of Circular and Spherical Higher-Order Ambisonics: Part II—Hybrid System