Audio Reproduction Research Articles

A study of the ultrasonic and audible frequency response of commercial loudspeakers for use as parametric acoustic arrays

Parametric acoustic loudspeakers (PAL), also known as parametric acoustic arrays (PAA), are speakers that use ultrasonic waves to generate directive audible sound. They emit an ultrasonic carrier modulated in amplitude by the desired audible signal, and due to non-linear propagation in air, this signal is reproduced in the audible range. Typical applications include audio reproduction for entertainment, but also active noise control and acoustic measurement of material properties. PAA typically consist of an array of piezoelectric transducers set on a surface. However, there are other speakers in the market that, due to their characteristics, could also achieve the PAL effect. This work examines the performance of commercial speakers when reproducing. First, the ultrasonic frequency response of each speaker is examined in free-field conditions using exponential sine sweeps (ESS). This enables searching for potential carrier frequencies to modulate the ESS in the ultrasonic range. The ESS are next modulated in the desired ultrasonic region and emitted through each speaker, obtaining their audible and ultrasonic impulse responses. The results show that some commercial speakers can also operate as PALs, which is interesting from an industrial point of view as it lowers costs and simplifies the assembly process.

Development of automatic audio panning system for immersive sound through stage size-aware model and object-tracking

We investigated a novel AI-assisted automated 'immersive' audio panning system designed to track audio-related objects within a video clip. This system comprises four sequential steps: Object-Tracking, Stage dimension Estimation, XY-Coordinate Calculation, and Object Audio Rendering. The system is designed to overcome existing challenges arising from the rapid and frequent movement of target objects by employing a pre-trained object-tracking model and integrating depth information to ensure stability in subsequent tasks. Additionally, we introduce a stage size-aware model to extrapolate stage dimension using our manually collected dataset, formatted as (Image, Width, Depth), which facilitates model training. Consequently, the system calculates XY-Coordinate pairs, serving as panning values for conventional audio mixers or decoders to enable immersive audio reproduction. We anticipate that this video- and space-aware automatic panning system will be valuable for the rapid production of new media.

Investigating the after effect of head-tracked movement on externalization for varying sound sources in binaural synthesis.

Binaural audio reproduction aims to create a realistic auditory experience for the listener through headphones. However, front and rear sound sources are often internalized (perceived inside the head) rather than externalized (perceived outside the head). Previous studies have shown that large head movements can significantly improve externalization, and that this improvement can persist even after head movements have stopped. This study examined how head-tracked movements influence the perception of sound externalization depending on the nature of the sound sources. The stimuli consisted of a first ``head movement" part during which the participants had to either perform a head movement or stay still, followed by a "rating" part where they evaluated externalization without moving. The sound sources during the ``rating" part could be similar, slightly different, or very different from the source during the head movement part. Results demonstrated that head-tracked movements effectively heightened externalization perception for all types of sources immediately after the movement. The type of source (female voice, male voice, or pink noise) did not affect perceived externalization. These findings suggest that the externalization enhancement induced by head-tracked movement persists after the movement ends even when encountering a different sound source.

Analysis and Model of Temporal Sound Attributes from Recorded Audio

A computational framework is proposed for analyzing the temporal evolution of perceptual attributes of sound stimuli. As a paradigm, the perceptual attribute of envelopment, which is manifested in different audio sound reproduction formats, is employed. For this, listener temporal ratings of the envelopment for mono, stereo, and 5.0-channel surround music samples, serve as the ground truth for establishing a computational model that can accurately trace temporal changes from such recordings. Combining established and heuristic methodologies, different features of the audio signals were extracted at each segment that envelopment ratings were registered, named long-term (LT) features. A memory LT computational stage is proposed to account for the temporal variations of the features through the duration of the signal, based on the exponentially weighted moving average of the respective LT features. These are utilized in a gradient tree boosting, machine learning algorithm, leading to a Dynamic Model that accurately predicts the listener’s temporal envelopment ratings. Without the proposed memory LT feature function, a Static Model is also derived, which is shown to have lower performance for predicting such temporal envelopment variations.

Perceptual Comparison of 3D Audio Reproduction With and Without Bottom Channels

Perceptual Comparison of 3D Audio Reproduction With and Without Bottom Channels

Reduced complexity for sound zones with subband block adaptive filters and a loudspeaker line array.

Sound zones are used to reproduce individual audio content to multiple people in a room using a set of loudspeakers with controllable input signals. To allow the reproduction of individual audio to dynamically change, e.g., due to moving listeners, changes in the number of listeners, or changing room transfer functions, an adaptive formulation is proposed. This formulation is based on frequency domain block adaptive filters and given room transfer functions. To reduce computational complexity, the system is extended to subband processing without cross-adaptive filters. The computational savings come from recognizing that sound zones consist of part-solutions which are inherently band limited, hence, several subbands can be ignored. To validate the theoretical findings, a 27-channel loudspeaker array was constructed, and measurements were performed in anechoic and reflective environments. The results show that the subband solution performs identically to a full-rate solution but at a reduced computational complexity.

Performance bounds for source localization using tetrahedral microphones

The Cramér-Rao Bound (CRB) quantifies the minimum variance on a given unbiased estimation parameter. It can be used to derive a lower bound on source localization error for arrays, of which there are several probabilistic models in the literature. However, existing work only considers source localization using arrays of omnidirectional sensors. This presentation describes a model of the CRB as a minimum variance estimation of localization using multiple tetrahedral microphones which have cardioid microphones. Tetrahedral microphones are commonly used to record soundfields for virtual audio reproduction. The proposed method uses the cross spectral density of signals in order to calculate the CRB. This allows for more signal parameters to be included in the model. The model of the CRB was validated with an experiment in an anechoic room with a single sound source. Experimental results show the CRB forms a lower bound at low signal-to-noise ratios.

A 87 dB SNR and THD+N 0.03% HiFi Grade Audio Preamplifier

This paper presents a significant contribution to high-fidelity audio technology through the comprehensive design and implementation of an advanced audio-grade preamplifier integrated with a high-performance digital-to-analog converter (DAC). The versatility of the digital section allows seamless integration of audio sources from Universal Serial Bus (USB), Bluetooth, or Sony-Philips Digital interface (SPDIF). In addition, the preamplifier accommodates various analog sources, offering flexibility to users. The unique architecture features an innovative resistor network-based DAC and precision components in the analog section, paving the way for achieving unparalleled audio fidelity. The measured distortion values are remarkably low, with a total harmonic distortion plus noise (THD+N) of only 0.03%. Furthermore, the system exhibits a signal-to-noise ratio (SNR) of 87 dB, comparable to industry benchmark. The hardware design sections went into meticulous details, providing an in-depth exploration of each block of the project. The inclusion of test and measurement data from the actual prototype enhances the credibility of the proposed work. This research not only showcases the theoretical operation of the preamplifier and DAC stage but also validates the practical implementation through many rounds of testing and validation. Our proposed work contributes to the field of high-fidelity audio by presenting a novel solution that is comparable to industry standards. The design’s adaptability to many different audio sources, coupled with robust performance metrics, positions it as a benchmark in the field of audio reproduction. The measurement results and findings presented in this paper are poised to influence future advancements in HiFi audio technology.

Spatial audio reproduction for studying second language speech perception in varying acoustic environments

This paper introduces recent attempts of utilising spatial audio reproduction in linguistic studies and discusses challenges of the new methodology. Thanks to its high reproducibility of arbitrary acoustic environments in a controlled laboratory setting, spatial audio reproduction has a potential to address the difficulties of examining participants with different language background that have to be recruited in geographically separated locations. A case study that investigated the difference in speech intelligibility in noise between first and second language listeners in varying acoustic environments is reported with its experimental results. The paper then attempts to verify the validity of the methodology using spatial audio reproduction by comparing experimental results conducted in real and virtual rooms.

Development and perceptual evaluation of hearing aids using interactive virtual environments

In hearing research, it has been shown that the realism of the test conditions has an influence on speech intelligibility thresholds, the subject's behavior, learning abilities and listening effort. This talk presents two multimodal experiment setups that aim to bridge the differences between the laboratory and reality, focusing on the development and perceptual evaluation of hearing devices. The first laboratory uses 360° video and higher order ambisonics stimuli, a head-mounted display, and a VR perceptual evaluation interface to asses the effect of hearing devices on noise intrusiveness, speech quality, and naturalness. The second laboratory is a large loudspeaker array inside of which subjects movements and behavior can be tracked and videos can be projected on walls. This talk will introduce several challenges encountered in the audio reproduction in these two laboratories and will also present a broad overview of solutions used and developed to ensure the validity of the experimental conditions.

Impact of direct-to-reverberation ratio and interaural cross-correlation on externalization in binaural audio rendering

To achieve realistic binaural reproduction of spatial audio in a headphone scenario, addressing inside-the-head-locatedness is crucial, which can be accomplished through appropriate externalization techniques. Two acoustic cues, the direct-to-reverberation ratio (DRR) and interaural cross-correlation (IACC), are known to influence perceived externalization. However, the precise relationship between these cues and their combined effect has not been extensively studied. In this investigation, we aim to explore the relative contribution of DRR and IACC to externalization by examining various combinations of these two acoustic cues. To this end, participants were exposed to various binaural room impulse responses (BRIRs) synthesized to produce three distinct levels of DRRs and IACCs for two different source directions. The listening test was conducted in an anechoic chamber with participants’ eyes blindfolded to minimize the influence of room divergence effect. We also investigated the influence of head tracking by conducting experiments with and without head trackers. The correlation maps between externalization and the two acoustic cues were generated for each source direction. The study reveals that the contribution of the two acoustic cues differs with and without head tracking, and the joint contribution also varies depending on the source direction and amount of reflections included in the BRIR.

Spatial audio reproduction of floor impact-borne sounds for auditory demonstrations in living environment

Floor impact-borne sound in the multistory residential buildings has been a serious social issue in Korea for a long time. The housing and land authority in a ministry of Korea recently amended the floor impact-borne sound evaluation method and criteria in response to the acoustic characteristics of the actual impact sources perceived in our daily lives. However, many people have not yet been convinced by the effectiveness of those evaluation method and criteria. This study discusses spatial audito reproduction of the various floor impact-borne sounds for auditory demonstrations to increase the laypersons’ understanding of the acoustic features of impact sources. The impact-borne sounds generated both by the standard impact sources used in evaluation and by some actual ones were recorded at a test-bed built in a typical apartment type. Their sound levels and reverberation times were also obtained across 1/3rd octave bands for reference. A multi-channel audio playback system engaging front/rear/ceiling speakers with subwoofers was integrated in a test-bed with a typical setting of living environment to reproduce a realistic sound field. It was optimized to create the sounds close to those actually heard in the room. The impact-borne sounds of the different damping floor systems will also be presented.

Spatial audio reproduction over ad hoc loudspeaker array using near-field compensation in spherical harmonics domain

Spatial audio reproduction using the loudspeaker array introduces the curvature effect leading to a distorted listening experience when the listener is in the near field. In the near-field, the loudspeakers are approximated as point sources (spherical wave) and amplify the mode vectors. Further, the problem becomes more challenging for the irregular loudspeaker arrangement, which causes uneven energy distribution in the reproduction region. In this context, a near-field compensation is applied to the encoded ambisonics coefficients. An optimization problem is formulated, such as the loudspeaker gains encoded with spherical harmonics basis coefficients should match the target ambisonics coefficients. Further, the in-phase and quadrature components of the energy localization vector are imposed as the constraints to direct maximum energy in the reproduction region. The solution to the optimization problem is obtained using a derivative-free optimization solver. The performance of the proposed methods is evaluated for ITU-R recommended loudspeaker layouts using the technical and perceptual evaluation attributes.

AsQM: Audio Streaming Quality Metric Based on Network Impairments and User Preferences

Audio streaming services have many users because the proliferation of cloud-based audio streaming services for different content. The complex networks that support these services do not always guarantee an acceptable quality on the end-user side. In this paper, the impact of temporal interruptions on the audio streaming reproduction and the user’s preference in relation to audio content are studied. In order to determine the key parameters in the audio streaming service, subjective tests are conducted, and their results show that a user’s quality-of-experience (QoE) is highly correlated with the following application parameters: the number of temporal interruptions or stalls, its frequency and length, and the temporal location where they occur. However, most important, the experimental results demonstrate that users’ preference for audio content plays an important role in users’ QoE. Thus, a preference factor (PF) function is defined and considered in the formulation of the proposed metric named audio streaming quality metric (AsQM). Considering that multimedia service providers are based on web servers, a framework to obtain user information is proposed. Furthermore, the results show that the AsQM implemented in the audio player of an end user’s device has a low impact on energy, processing and memory consumption.

Direct acoustic velocity vector measurements for evaluation of multichannel sound reproduction systems with signals of varied partial coherence

In multichannel sound reproduction systems, the perceived virtual auditory scene is strongly influenced by inter-channel phase-shifts and the degree of mutual coherence among channel signals. Such degrees of freedom may be employed to effect an expanded virtual sound stage or a sense of spaciousness, but they also may lead to unwanted spectral coloration, diffuse or ambiguous virtual source locations, or startling and disorienting spatial image perceptions often described as “phaseyness.” Such effects may be described and understood in terms of the acoustic velocity in the vicinity of the listener. In this work we present theoretical predictions of the acoustic velocity vector for a stereo reproduction system with channel signals of varying phase and levels of mutual coherence, and we report direct measurements of the acoustic velocity vector employing a 3D sound intensity probe for several specific cases. Acoustic velocity vector predictions and measurements also are correlated with listener perceptions. The presented measurements are also compared to subjective localization of sound sources. The direct measurement of the acoustic velocity vector in spatial sound reproduction systems is a useful tool to quantitatively evaluate the performance of multi-channel spatial audio reproduction systems, to predict listener perceptions of the acoustic field reproduced by such systems, and to investigate the impact of early reflections in a room on localization.

Disentangling the Rhythm from the Melody: What Really Causes the Mozart Effect?

The Mozart Effect refers to enhanced spatial ability following listening to 10 minutes of Mozart music (Rauscher et al., 1993; Rauscher et al., 1995; Rideout & Taylor, 1997). However, many researchers have not been able to replicate its benefit on cognitive functioning (McKelvie & Low, 2002; Steele et al., 1999) while other researchers provide evidence that the intellectual benefit may not be due to the music exclusively, but to changes in mood and arousal levels (Cacciafesta et al., 2010; Steele, 2000; Lints & Gadbois, 2003; Thompson et al., 2001). Furthermore, it is unclear whether this Mozart Effect is the result of the melody or the rhythmic pattern of the Mozart music (Shi, 2020). Therefore, the purpose of the current study was to determine which musical element (i.e., melody or rhythmic pattern) of Mozart’s sonata (Mozart Sonata for Two Piano’s in D major, K. 448) causes the spatial performance enhancement and to determine if the benefit generalizes to another type of spatial ability, specifically 2D mental rotation. In order to disentangle the effects of the rhythm from the melody, we exposed participants to a Lo-fi (i.e., reproduction of audio that highlights the rhythmic pattern of Mozart’s sonata with a slower tempo) version of the Mozart sonata that maintained the rhythmic pattern of the sonata but differed in melody. Inconsistent with hypotheses, neither version of the Mozart music significantly improved mood and arousal; however, the mood/arousal benefits approached significance while the control music, Albinoni’s Adagio in G Minor for Organ and Strings, significantly decreased mood/arousal. Thus, the current results provide some evidence that the rhythmic pattern (as opposed to the melody) of the Mozart sonata increases mood/arousal. However, the rhythmic pattern failed to produce better cognitive functioning (e.g., faster 2D mental rotation) in either its original form or the Lo-fi form.

Sounding the Text: Susan Howe and David Grubbs’s Thiefth

This essay investigates the musico-poetic collaboration between Susan Howe and David Grubbs. Focusing on Thiefth (2005), the first of a CD series that collects Thorow (1987; 1990) and Melville’s Marginalia (1993), it examines the forms, effects, and implications of the two works’ remediation from printed to sonic product. At stake are notions of textuality and voice, as well as issues of margins and marginalization that Howe starts challenging on page. The shift from the written to the aural dimension does in fact intensify the poems’ antirepresentative and anti-narrative drive, extending Howe’s dismantling of the visual frames. As the poet’s selective, anti-expressivist and anti-performative reading increases her textual scattering and fragmentation, foregrounding the sonic and material aspects of language, so does Grubbs’s experimentation with music, sound, and voice manipulation through the use of audio reproduction technology. If the insertion of pre-recorded ambient sounds generates acoustic effects that match the polyvocality and simultaneity of Howe’s visual poems, music amplifies their inherent dissonance and release from the constraints of signification. This intricate web of sonorities does not only defy the authority, stability, and closure of the written texts. It develops an aesthetics of sound that augments Howe’s graphic experimentation and calls for a listening practice that might draw attention to the margins of history and society.

Hybrid controller for mid-range mid-power audio application

A major reason for compromised audio quality with a coil-type loudspeaker is its complex impedance profile, which disturbs the intended algebraic linearity or linear phase behavior between the amplifier input and speaker driving force. Though switch-mode amplifiers are increasingly preferred in such audio systems owing to their superior energy efficiency, their inherent non-linearity and finite internal impedance further degrade audio reproduction. This paper investigates the effect of stated impedance on the loudspeaker driving current, hence on the loudspeaker driving force, and proposes a Hybrid controller-amplifier to enhance the proportionality between the audio reference and driving force without compromising on loudspeaker damping. The Hybrid amplifier combines a sliding mode controlled amplifier (SMCA) and a linear controller, designed after exhaustive modeling of the input–output behavior of the SMCA and best-fit approximation of the loudspeaker impedance. An 8 W, 500 Hz to 20 kHz prototype of the proposed amplifier is fabricated on the basis of the presented design. Time and frequency-domain results obtained with the experimental hardware are presented to validate the performance improvement.

Selecting a reference headphone for 3D audio reproduction

3D audio uses generalized or personalized head-related-transfer-functions (HRTF) to provide the essential spatial and timbral cues. The headphone must reproduce these cues without introducing any errors of its own. A standard approach is to first calibrate the headphone by measuring and equalizing its frequency response at the blocked ear canal of the listener, or in the generalized case, a standard test fixture. However, the reliability of the calibration can be adversely affected by acoustical interactions between the headphone, the listener or the test fixture that can vary with headphone model, individual listener, and how the headphone is placed on their ears. The extent to which these errors affect the quality of 3D audio reproduction is not well understood. This paper discusses some recent investigations into the acoustical interactions between headphones, listeners, and standard test fixtures, and the effect on the quality of 3D audio reproduction through headphones. Several different models of headphones were measured on both human subjects and test fixtures with multiple placements to quantify the errors and their reproducibility. From these data, the headphone with the most linear and consistent response was selected for 3D audio reproduction.

The Soundlapse Project: Exploring Spatiotemporal Features of Wetland Soundscapes

Abstract This article discusses an interdisciplinary project aimed at highlighting the acoustical heritage of wetlands by means of field recordings and a novel time-lapse montage method. The authors discuss a site-specific sound installation that was designed using original wetlands field recordings, live processing and spatial audio multichannel reproduction. The discussion focuses on spatial and temporal features of different types of recorded wetlands soundscapes. Future developments of this project will consider the implementation of a standalone spatiotemporal application, to be used in the context of virtual reality applications, game audio and interactive dance performance.