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Pioneered in the sixties, Crosstalk Cancellation (CTC) allows for immersive sound reproduction from a limited number of loudspeakers. Upcoming virtual reality and augmented reality applications, as well as widespread availability of 3D audio content, have boosted interest in CTC technologies over the recent years. In this paper, we present a novel multiband approach to CTC, evolving and superseding our original work based on modeling of the system’s geometrical acoustics. This new solution, whilst keeping a simple processing model, offers improved CTC effectiveness, reduced residual coloration and wider bandwidth. The enhanced performance of our new approach has been confirmed by laboratory experiments.
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The frequency response of a headphone is very important for listener satisfaction. Listener preferences have been thoroughly studied for frequencies below 10 kHz, but preferences above that frequency are less well known. Recent improvements in the high frequency performance of ear simulators make it more practical to study this frequency region now. In this study, the high frequency portion of the target response for insert headphone designs is investigated, focusing on the range from 10 kHz to 20 kHz. A new target response is proposed, based on listener preference ratings in a blind listening test. The results show a clear preference for significantly more high frequency energy than was proposed in a previous popular earphone target curve. The preferred response is also affected by the listener’s hearing thresholds, with additional high frequency boost being preferred for listeners with age-related hearing loss.
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With the advent of Web Audio the compliant browser offers a toolbox of audio production components right out of the box. This could prove valuable for any content producer in the audio field, either professional or amateur. In this paper we will add deep learning to this scenario with the ultimate goal of obtaining machine assisted real-time audio production in-browser. As a proof of concept we will implement a basic yet complete one-channel design that uses deep learning to assist an automatic filtering algorithm producing parameters to adjust an audio signal in the Web Audio context of a browser. To achieve this we will evaluate five-class audio prediction models and compare their accuracy at the model building stage with their accuracy when exported to the real-time context. We will present two ways to measure this accuracy in real-time. We will also present a method to reduce jumpiness in real-time predictions when classification scores are ambiguous. We will highlight some important limitations and we will also present a refined model – designed for our domain specific audio set – based on some architectures from previous research and find that that our model outperforms these architectures
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Sound system designers are used to optimizing loudspeaker systems for the audience experience with free-field simulation software. However, noise pollution reduction must also be considered during the design phase and the propagation of sound may be affected by inhomogeneous atmospheric conditions, such as wind, temperature gradients, and ground impedance. This paper proposes a method to simulate the impact of the environment on sound pressure levels at large distances created by loudspeaker systems using parabolic equations, considering a reference left-right main system associated with either flown or ground-stacked subwoofers. Results show a higher variability of the sound pressure level with systems using ground-stacked subwoofers. The influence of the crossover frequency between main and subwoofers is discussed in this paper.
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Piezoelectric actuators offer advantages in terms of weight and form-factor when compared with traditional inertial exciters used to drive flat-panel loudspeakers. Unlike inertial exciters, piezoelectric actuators induce vibrations in the panel by generating bending moments at the actuator edges. Models for piezoelectric excitation are developed, and it is shown that piezoelectric actuators are most effective at driving resonant modes whose bending half-wavelength is the same size as the actuator dimensions, giving a natural boost to the high-frequency response. Flat-panel loudspeaker design techniques such as the modal crossover method, and the corresponding array layout optimization are adapted using the model for the response of piezo-driven panels. The adapted design techniques are shown to eliminate the isolated, low-frequency bending modes responsible for reduced sound quality on a prototype panel speaker.
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When combining close microphone(s) with ambience microphone(s), the direct signal will be picked up slightly delayed by the ambience microphone(s) compared to the close microphone(s). This study examines the auditory significance of the delayed direct signal in the context of early reflections and reverberation emitted from the boundaries within an acoustic environment. The aim is to establish to which extent any auditory perception influenced by the delayed direct signal may impact the application of the ambience microphone setup itself. The findings in this study suggests that when combined with close microphones, the ambience setup when using omni microphones, should be placed at a distance beyond at least twice the critical distance of the acoustical recording environment relative to the sound source.
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Current systems capable of 3D sound reproduction and panning of multi-channel sound sources require the user to employ standard, low-level interaction tools such as knobs and sliders to handle audio sources and reproduction setups with higher complexity. Innovative tools and control methods designed to meet the requirements of immersive audio environments are essential to enable efficient implementation of these technologies and software. This work investigated three types of input devices as controllers for an immersive audio mixing task in a spatial audio system. This second part of the research focuses on the objective data registered by the apparatus of the experiment. Results suggest devices with a high degree of integrality had the fastest learning curve and produced comparable results to the traditional method in response time.
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Networked Music Performances (NMP) are of increasing importance for musical enthusiasts, amateurs and professionals exploring new solutions and opportunities to rehearse or perform together at geographically distant locations. The use of public cellular connectivity to access wide area networks for such purposes can provide unique flexibility in planning NMP setups. Generally, latency and reliability are two key parameters for the end-to-end transmission of audio information through networks in this context. The stringent requirements of NMPs with respect to those parameters pose a major challenge for the current fourth generation (4G) of cellular technology. It is expected that the new and upcoming fifth generation (5G) of cellular technology will deliver significant Key Performance Indicator (KPI) improvements, and thus could be a promising enabler for musical interaction over distant locations. Currently, first public deployments of cellular 5G are being rolled out. This work presents an in-depth latency and reliability analysis of a distributed performance conducted over public 5G infrastructure in Finland. Measurement results suggest that Quality-of-Service (QoS) is needed in order to enable NMP over cellular 5G for many users and devices in a consistent, plannable and also flexible way.
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This paper presents analysis methods that make use of synthesis concepts by means of movements of sounds in virtual 3D auditory space at audio rate. The main focuses lies on the modulating effect caused by moving a sound around a listener, commonly referred to as panning. Here, we will demonstrate a comparison between two common panning approaches: Vector-Base Amplitude Panning (VBAP) and Wavefield Synthesis (WFS). Both apply some type of secondary source selection process, reducing the number of active loudspeakers at any moment in time. Under specific conditions it therefore becomes possible to compare VBAP and WFS directly. Using a technique originally developed for 3D sound synthesis over multichannel loudspeaker arrays at audio rate, both VBAP and WFS are adapted to be emulated by this technique. By using this sonification process, the differences between panning with VBAP or WFS can then not only be visualised but also heard. This makes spatial sound synthesis not just a creative tool for sound creation, but also an analytic tool that can help illustrate characteristics of spatialisation approaches.
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Size and design constraints in products such as soundbars and TVs require loudspeaker spiders of small diameter to allow for large voice-coil excursion. Spider designs that undergo exceedingly large displacements can exhibit buckling of the spider rolls, resulting in very audible distortion. Such buckling events are non-trivial to simulate with finite element methods and often lead to solver non-convergence. When wrapping numerical optimization algorithms around the finite-element simulations to achieve optimal spider designs, it is important to ensure that all simulated designs can be solved without errors or convergence issues. The optimal spider design might be right within the buckling limits and an automated numerical optimization algorithm will need to be able to resolve some designs that exhibit buckling. This work shows how an augmented FEM method can be used to circumvent issues when employing numerical optimization for a spider design near its buckling limits.
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