Perceptual Sound Field Reconstruction and Coherent Emulation

该项目涉及开发一种新的 5--10 通道音频技术，该技术将在以下方面改进现有技术：(a) 真实感、(b) 听觉视角的准确性和稳定性、(c) 最佳听音点的大小以及 (d) 包围体验。由于新技术旨在创建 360 度听觉视角，因此再现将通过位于正多边形顶点的扬声器进行。每个扬声器将由两个组件组成，一个组件向听众辐射直接声场，另一个组件通过引入额外的散射来再现扩散声场。下面列出的特定任务的目标是找到捕获声场线索的最佳方法，并使用所提出的回放系统以提供原始或所需声场最令人信服的幻觉的方式渲染它们。（i）将研究所提出的回放系统的最佳麦克风阵列。所考虑的阵列将由放置在水平面中正多边形顶点处的麦克风组成，麦克风的数量等于扬声器的数量。对于每个阵列，将考虑从接近重合到稍微超出最佳值的不同直径以及不同的麦克风方向性图案。这些研究将针对扬声器配置的几个直径进行重复，以研究最佳阵列直径是否取决于扬声器布局的尺寸，如果是，则表征这种依赖性。还将研究和表征最佳麦克风方向性模式和阵列直径之间可能的依赖性。将根据上述标准 (a)-(d) 在关键听力测试中对阵列进行评估。实验将以模拟为指导，模拟将对聆听区域内生成的 ITD 和 ILD 线索进行初步客观评估。同时，还将研究所提出的技术生成的声场的数学模型，这可以为最佳麦克风阵列设计提供一些额外的见解。 (ii) 将系统地研究消除串扰的回放的影响。将首先使用现有的串扰消除算法，如有必要，将开发在一系列收听环境中具有数值效率和效果的新算法。然后将研究用于具有串扰消除的回放的最佳麦克风阵列，即，将重复(i)下描述的工作以用于具有串扰消除的再现。最后，将比较具有和不具有串扰消除的最佳系统。(iii)将研究直接/漫射声场分离的算法。当乐器数量不超过麦克风数量时，可以使用多通道均衡技术找到干源信号，然后将其与房间脉冲响应的直达/混响部分进行卷积，分别获得直达/漫射声场分量。然而，由于脉冲响应过长，音频中的多通道均衡尤其具有挑战性，我们将为音频应用的多通道均衡开发数值高效的算法。然后我们将研究直接/扩散声场分解的心理声学近似，对声源的数量没有限制。 (iv) 将根据标准 (a)-(d) 在关键听力测试中系统地研究用于获取直接声场线索的近重合定向麦克风阵列和用于获取扩散声场线索的基于全向或双向麦克风的宽间隔阵列的组合。该方法将与 (i)-(iii) 中描述的方法进行比较，其中相同的阵列用于两个声场分量。

The project is concerned with the development of a new 5--10 channel audio technology which would improve over existing ones in terms of (a) realism, (b) accuracy and stability of the auditory perspective, (c) size of the sweet spot, and (d) the envelopment experience. Since the new technology aims to create a 360 degrees auditory perspective, the reproduction will take place over speakers positioned at vertices of a regular polygon. Each speaker will consist of two components, one which will radiate the direct sound field toward a listener, and another which will reproduce diffuse sound field by introducing additional scattering. The goal of the particular tasks, listed below, is to find optimal ways to capture sound field cues and render them using the proposed playback system in a manner which would provide the most convincing illusion of the original or desired sound field.(i) Optimal microphone arrays for the proposed play-back system will be investigated. Arrays considered will consist of microphones placed in the horizontal plane at the vertices of a regular polygon, with the number of microphones equal to the number of speakers. For each array, different diameters, in the range from near coincident up to somewhat beyond the optimal value, and different microphone directivity patterns will be considered. These studies will be repreated for a few diameters of the speaker configuration to investigate if the optimal array diameter depends on the size of the speaker lay-out, and if so to characterize that dependence. Possible dependencies between the optimal microphone directivity patterns and array diameters will be also investigated and characterized. Arrays will be evaluated in critical listening tests according to criteria (a)--(d) stated in the above. Experiments will be guided by simulations which would provide initial objective assessment of ITD and ILD cues generated within the listening area. In parallel, mathematical models of sound fields generated by the proposed technology will be investigated, which could provide some additional insight into the optimal microphone array design. (ii) The impact of play-back with cross-talk cancellation will be be systematically investigated. Existing cross-talk cancellation algorithms will be first used, and if necessary, new algorithms which are numerically efficient and effective in a range of listening environments will be developed. Then optimal microphone arrays for play back with cross-talk cancellation will be investigated, i.e. the work described under (i) will be repeated for reproduction with cross-talk cancellation. Finally, optimal systems with and without cross-talk cancellation will be compared.(iii) Algorithms for direct/diffuse sound field separation will be studied. When the number of instruments does not exceed the number of microphones, multichannel equalization techniques can be used to find dry source signals, which can then be convolved with direct/reverberant parts of room impulse responses to obtain direct/diffuse sound field components, respectively. Multichannel equalization in audio is, however, particularly challenging owing to excessively long impulse responses, and we will develop numerically efficient algorithms for multichannel equalization for audio applications. Then we will study psychoacoustic approximation to direct/diffuse sound field decomposition with no restriction on the number of sources. (iv) Combinations of near-coincident directional microphone arrays, for acquiring direct sound field cues, and widely spaced arrays based on omni-directional or bi-directional microphones, for acquiring diffuse sound field cues, will be systematically investigated in critical listening tests according to criteria (a)--(d). This approach will be evaluated in comparison with the approach described in (i)--(iii) where the same array is used for both sound field components.