Authors:Blanco Galindo, Miguel; Coleman, Philip; Jackson, Philip J. B.
Affiliation:Centre for Vision, Speech and Signal Processing, University of Surrey, Guildford, UK; Institute of Sound Recording, University of Surrey, Guildford, UK
Microphone array beamforming can be used to enhance and separate sound sources, with applications in the capture of object-based audio. Many beamforming methods have been proposed and assessed against each other. However, the effects of compact microphone array design on beamforming performance have not been studied for this kind of application. This study investigates how to maximize the quality of audio objects extracted from a horizontal sound field by filter-and-sum beamforming, through appropriate choice of microphone array design. Eight uniform geometrieswith practical constraints of a limited number of microphones and maximum array size are evaluated over a range of physical metrics. Results show that baffled circular arrays outperform the other geometries in terms of perceptually relevant frequency range, spatial resolution, directivity, and robustness. Moreover, a subjective evaluation of microphone arrays and beamformers is conducted with regards to the quality of the target sound, interference suppression, and overall quality of simulated music performance re- cordings. Baffled circular arrays achieve higher target quality and interference suppression than alternative geometries with wideband signals. Furthermore, subjective scores of beamformers regarding target quality and interference suppression agree well with beamformer on-axis and off-axis responses; with wideband signals, the superdirective beamformer achieves the highest overall quality.
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Authors:McCormack, Leo; Pulkki, Ville; Politis, Archontis; Scheuregger, Oliver; Marschall, Marton
Affiliation:Aalto University, Espoo, Finland; Technical University of Denmark, Kongens Lyngby, Denmark; Tampere University, Tampere, Finland
This article details an investigation into the perceptual effects of different rendering strategies when synthesizing loudspeaker array room impulse responses (RIRs) using microphone array RIRs in a parametric fashion. The aim of this rendering task is to faithfully reproduce the spatial characteristics of a captured space, encoded within the input microphone array RIR (or the spherical harmonic RIR derived from it), over a loudspeaker array. For this study, a higherorder formulation of the Spatial Impulse Response Rendering (SIRR) method is introduced and subsequently employed to investigate the perceptual effects of the following rendering configurations: the spherical harmonic input order, frequency resolution, and utilizing ded- icated diffuse stream rendering. Formal listening tests were conducted using a 64-channel loudspeaker array in an anechoic chamber, where simulated reference scenarios were compared against the outputs of different methods and rendering con- figurations. The test results indicate that dedicated diffuse stream rendering and higher analysis orders both yield noticeable perceptual improvements, particularly when employing problematic transient stimuli as input. Additionally, it was found that the frequency resolution employed during rendering has only a minor influence over the perceived accuracy of the reproduc- tion in comparison to the other two tested attributes.
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Authors:Tian, Xing; Shen, Yong; Chen, Li; Zhang, Zhaoqi
Affiliation:Key Laboratory of Modern Acoustics, Institute of Acoustics, Nanjing University, Nanjing, China; Shenzhen Research Institute of Nanjing University, Shenzhen, China
Fractional derivative loudspeaker model describes the physical mechanism of eddy current losses in the voice coil and visco- elasticity of the suspension, which simplifies existing lumped parameter models containing creep and inductance corrections and makes more accurate predictions of loudspeaker dynamics. However, research on nonlinear fractional derivative model stays at the simulation level due to the difficulty in measuring the nonlinear parameters. In this paper, an identification algo- rithm is proposed to address this problem. All model parameters including four nonlinear parameters can be precisely identi- fied based on two small-signal characteristic curves and electrical data under large-signal conditions. Objective experimental indicators confirm the validity of the proposed algorithm, and the reason why nonlinear fractional derivative model performs better is analyzed.
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Authors:Gabrielli, Leonardo; Tomassetti, Stefano; Squartini, Stefano
Affiliation:Università Politecnica delle Marche, Ancona, Italy
In many sound synthesis applications, once the salient properties of a sound source are defined, equalization is undertaken in order to fit it to the musical context. This is normally done by linear filters.When the sound synthesis algorithm, however, is as flexible as a physical model, Computational Sound Design techniques can be employed to perform offline equalization at the model level, saving real-time computational resources. In this work we extend the use of a recently proposed algorithm, MORIS, to equalize the harmonic content of a synthesized sound obtained by physical modeling according to equalization curves. We evaluate the effectiveness of the approach by considering an open-source clarinet model and pipe organ physical model, comparing our results with those obtained by online filtering and showing the advantages of the new method.
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Authors:Cui, Linlin; Cui, Weili
Affiliation:Zhongyuan University of Technology
At present, the most commonly used single arrays for acoustic localization consist of four, five, and six-element arrays as well as circle arrays. This acoustic localization model based on an integrated array is proposed to solve the inherent defects on a single array. The integrated array consists of an inner solid four-element array and outer plane five-element array. Utilizing the principle of TDOA, the inside solid four-element array is used for calculating the azimuth and pitch angles. The outside plane five-element array is used for calculating the distance. With fireworks as the sound sources, five sound source points with a 200 m by 200 m area are selected at random in a Beijing suburb in China. The experimental results indicate that the localization errors are all less than 0.6 m.
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When it comes to evaluating advanced spatial audio or immersive audio technology there may be advantages to adopting some form of reference-free evaluation. There’s also the question of defining immersion carefully enough, and a comprehensive attempt at doing this in the audio context has now been undertaken. We find that non-experienced listeners may not have the same preference patterns as experienced ones in this context, and that additional low-slung loudspeakers can improve the results in certain cases. Finally, the potential of soundbar technology is considered.
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