scholarly journals Spatial Multizone Soundfield Reproduction Design

2021 ◽  
Author(s):  
◽  
Wenyu Jin

<p>It is desirable for people sharing a physical space to access different multimedia information streams simultaneously. For a good user experience, the interference of the different streams should be held to a minimum. This is straightforward for the video component but currently difficult for the audio sound component. Spatial multizone soundfield reproduction, which aims to provide an individual sound environment to each of a set of listeners without the use of physical isolation or headphones, has drawn significant attention of researchers in recent years. The realization of multizone soundfield reproduction is a conceptually challenging problem as currently most of the soundfield reproduction techniques concentrate on a single zone.  This thesis considers the theory and design of a multizone soundfield reproduction system using arrays of loudspeakers in given complex environments. We first introduce a novel method for spatial multizone soundfield reproduction based on describing the desired multizone soundfield as an orthogonal expansion of formulated basis functions over the desired reproduction region. This provides the theoretical basis of both 2-D (height invariant) and 3-D soundfield reproduction for this work. We then extend the reproduction of the multizone soundfield over the desired region to reverberant environments, which is based on the identification of the acoustic transfer function (ATF) from the loudspeaker over the desired reproduction region using sparse methods. The simulation results confirm that the method leads to a significantly reduced number of required microphones for an accurate multizone sound reproduction compared with the state of the art, while it also facilitates the reproduction over a wide frequency range.  In addition, we focus on the improvements of the proposed multizone reproduction system with regard to practical implementation. The so-called 2.5D multizone oundfield reproduction is considered to accurately reproduce the desired multizone soundfield over a selected 2-D plane at the height approximately level with the listener’s ears using a single array of loudspeakers with 3-D reverberant settings. Then, we propose an adaptive reverberation cancelation method for the multizone soundfield reproduction within the desired region and simplify the prior soundfield measurement process. Simulation results suggest that the proposed method provides a faster convergence rate than the comparative approaches under the same hardware provision. Finally, we conduct the real-world implementation based on the proposed theoretical work. The experimental results show that we can achieve a very noticeable acoustic energy contrast between the signals recorded in the bright zone and the quiet zone, especially for the system implementation with reverberation equalization.</p>

2021 ◽  
Author(s):  
◽  
Wenyu Jin

<p>It is desirable for people sharing a physical space to access different multimedia information streams simultaneously. For a good user experience, the interference of the different streams should be held to a minimum. This is straightforward for the video component but currently difficult for the audio sound component. Spatial multizone soundfield reproduction, which aims to provide an individual sound environment to each of a set of listeners without the use of physical isolation or headphones, has drawn significant attention of researchers in recent years. The realization of multizone soundfield reproduction is a conceptually challenging problem as currently most of the soundfield reproduction techniques concentrate on a single zone.  This thesis considers the theory and design of a multizone soundfield reproduction system using arrays of loudspeakers in given complex environments. We first introduce a novel method for spatial multizone soundfield reproduction based on describing the desired multizone soundfield as an orthogonal expansion of formulated basis functions over the desired reproduction region. This provides the theoretical basis of both 2-D (height invariant) and 3-D soundfield reproduction for this work. We then extend the reproduction of the multizone soundfield over the desired region to reverberant environments, which is based on the identification of the acoustic transfer function (ATF) from the loudspeaker over the desired reproduction region using sparse methods. The simulation results confirm that the method leads to a significantly reduced number of required microphones for an accurate multizone sound reproduction compared with the state of the art, while it also facilitates the reproduction over a wide frequency range.  In addition, we focus on the improvements of the proposed multizone reproduction system with regard to practical implementation. The so-called 2.5D multizone oundfield reproduction is considered to accurately reproduce the desired multizone soundfield over a selected 2-D plane at the height approximately level with the listener’s ears using a single array of loudspeakers with 3-D reverberant settings. Then, we propose an adaptive reverberation cancelation method for the multizone soundfield reproduction within the desired region and simplify the prior soundfield measurement process. Simulation results suggest that the proposed method provides a faster convergence rate than the comparative approaches under the same hardware provision. Finally, we conduct the real-world implementation based on the proposed theoretical work. The experimental results show that we can achieve a very noticeable acoustic energy contrast between the signals recorded in the bright zone and the quiet zone, especially for the system implementation with reverberation equalization.</p>


2021 ◽  
Vol 263 (6) ◽  
pp. 335-341
Author(s):  
Xi Hong ◽  
Xiangyang Zeng ◽  
DU Bokai

Sound field reproduction aims to create or reproduce a desired sound environment, where both the audio content and the spatial property of the sound field are preserved. For a practical reproduction system which is usually placed in a real 'listening room', acoustic transfer function measurement of the loudspeaker array is a time consuming work. The equivalent source method is an option to interpolate loudspeaker array acoustic transfer functions over the target region in reverberant sound field and has been implemented in the preceding researches. However, the selection of the optimized distances of the equivalent sources remains a challenging problem, especially considering the complex acoustic environment in reverberant room. In this work, we apply a multilayer equivalent source method. A simulation is conducted in virtual listening rooms with different reverberation conditions to investigate the reproduction performance of the proposed method. The comparison with the conventional single layer equivalent source method is provided.


2014 ◽  
Vol 6 (6) ◽  
pp. 611-618 ◽  
Author(s):  
Yung-Wei Chen ◽  
Hung-Wei Wu ◽  
Yan-Kuin Su

In this paper, a new multi-layered triple-passband bandpass filter using embedded and stub-loaded stepped impedance resonators (SIRs) is proposed. The filter is designed to have triple-passband at 1.8, 2.4, and 3.5 GHz. The 1st and 2nd passbands (1.8/2.4 GHz) are simultaneously generated by controlling the impedance and length ratios of the embedded SIRs (on top layer). The 3rd passband (3.5 GHz) is generated by using the stub-loaded SIR (on bottom layer). Using the embedded SIR, the even modes can be tuned within very wide frequency range and without affecting the odd modes. Therefore, the design of multi-band filters with very close passbands can be easily achieved and having a high isolation between the passbands. The filter can provide the multi-path propagation to enhance the frequency response and achieving the compact circuit size. The measured results are in good agreement with the full-wave electromagnetic simulation results.


Micromachines ◽  
2021 ◽  
Vol 12 (2) ◽  
pp. 168
Author(s):  
Xiaoyong Zhang ◽  
Guojun Zhang ◽  
Zhenzhen Shang ◽  
Shan Zhu ◽  
Peng Chen ◽  
...  

The principle of acoustic energy flux detection method using a single micro electromechanical system (MEMS) vector hydrophone is analyzed in this paper. The probability distribution of acoustic energy flux and the weighted histogram algorithm are discussed. Then, an improved algorithm is proposed. Based on the algorithm, the distribution range of the energy is obtained by a sliding window, the energy center of gravity in the range is considered as the result of direction of arrival (DOA) estimation, and it is proved to be the maximum likelihood estimation of the target direction. The simulation results show that, with the signal to noise ratio (SNR) from −10 dB to 10 dB, the root mean square error (RMSE) of the improved algorithm is reduced by 47.8% on average, and is more accurate in the presence of interference. The experimental results of lake test are consistent with the theory analysis and simulation results.


Author(s):  
Tomoko Ohya ◽  
◽  
Masanori Shimamoto ◽  
Takayuki Shiose ◽  
Hiroshi Kawakami ◽  
...  

This paper proposes a framework for analyzing the relationships between human recognition and the ecological aspects of physical space, aiming to support conflict management of architectural designs. Our framework employs novel fuzzy sets: the Constraint-Interval Fuzzy Set (CoIFS) [1], which visualizes roles of architectural spaces from the viewpoint of space usages and the physical abilities of target users. For designing indoor spatial layouts, the users are the inhabitants. From an ecological viewpoint, environments affect inhabitant behaviors on every location and scene. Among several ecological factors, we adoptpathway likenessandsettling attractionas examples of metrics for evaluating space. Inhabitants’ personalities are implemented by configurations of private/official space and settling attractions for simulating their behavior in a room. Based on simulation results, a spatial layout is evaluated by metrics that enable us to translate the roles of space into CoIFS. The potential of our framework is also discussed with the architectural notions ofP-SpaceandN-Space.


2008 ◽  
Vol 2008 ◽  
pp. 1-19 ◽  
Author(s):  
David Birchfield ◽  
Harvey Thornburg ◽  
M. Colleen Megowan-Romanowicz ◽  
Sarah Hatton ◽  
Brandon Mechtley ◽  
...  

We present concurrent theoretical work from HCI and Education that reveals a convergence of trends focused on the importance of three themes: embodiment, multimodality, and composition. We argue that there is great potential for truly transformative work that aligns HCI and Education research, and posit that there is an important opportunity to advance this effort through the full integration of the three themes into a theoretical and technological framework for learning. We present our own work in this regard, introducing the Situated Multimedia Arts Learning Lab (SMALLab). SMALLab is a mixed-reality environment where students collaborate and interact with sonic and visual media through full-body, 3D movements in an open physical space. SMALLab emphasizes human-to-human interaction within a multimodal, computational context. We present a recent case study that documents the development of a new SMALLab learning scenario, a collaborative student participation framework, a student-centered curriculum, and a three-day teaching experiment for seventy-two earth science students. Participating students demonstrated significant learning gains as a result of the treatment. We conclude that our theoretical and technological framework can be broadly applied in the realization of mixed reality, student-centered learning environments.


2012 ◽  
Vol 2012 ◽  
pp. 1-13 ◽  
Author(s):  
Jussi Rämö ◽  
Vesa Välimäki

Augmented reality audio (ARA) combines virtual sound sources with the real sonic environment of the user. An ARA system can be realized with a headset containing binaural microphones. Ideally, the ARA headset should be acoustically transparent, that is, it should not cause audible modification to the surrounding sound. A practical implementation of an ARA mixer requires a low-latency headphone reproduction system with additional equalization to compensate for the attenuation and the modified ear canal resonances caused by the headphones. This paper proposes digital IIR filters to realize the required equalization and evaluates a real-time prototype ARA system. Measurements show that the throughput latency of the digital prototype ARA system can be less than 1.4 ms, which is sufficiently small in practice. When the direct and processed sounds are combined in the ear, a comb filtering effect is brought about and appears as notches in the frequency response. The comb filter effect in speech and music signals was studied in a listening test and it was found to be inaudible when the attenuation is 20 dB. Insert ARA headphones have a sufficient attenuation at frequencies above about 1 kHz. The proposed digital ARA system enables several immersive audio applications, such as a virtual audio tourist guide and audio teleconferencing.


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