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Computational strategy for accelerating robust sound source detection in dynamic scenes

机译:在动态场景中加速鲁棒声源检测的计算策略

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Efficient sound source detection and location with microphone arrays is important for many applications, including teleconferencing, surveillance, and smart rooms. While the steered response power algorithms exhibit robust performance relative to other approaches, their applications are limited by the high computational load required. For dynamic auditory scenes, the entire space must be scanned at regular intervals due to moving sound sources switching between active and inactive states. This paper introduces a time segmentation and parallelization strategy to speed up the steered response power algorithm for dynamic auditory scenes with multiple speech sources. The primary application targeted by this work is for immersive arrays and off-line auditory scene analysis with beamforming for speaker separation in cocktail party environments. Results from a Monte Carlo simulation with 6 speech sources in a mildly reverberant environment demonstrate a speed-up factor of 45, with a modest loss in the number of detections and a significant reduction in anomalous detections. Experimental results with real recordings demonstrate a performance consistent with those of the simulation.
机译:麦克风阵列的高效声源检测和定位对于许多应用(包括电话会议,监视和智能室)很重要。尽管转向响应功率算法相对于其他方法表现出强大的性能,但其应用受到所需的高计算负荷的限制。对于动态听觉场景,由于移动声源在活动状态和非活动状态之间切换,因此必须按规则的间隔扫描整个空间。本文介绍了一种时间分割和并行化策略,以加快具有多个语音源的动态听觉场景的转向响应功率算法。这项工作的主要应用是用于沉浸式阵列和带有波束形成的离线听觉场景分析,用于在鸡尾酒会环境中分离扬声器。在轻微混响环境中使用6个语音源进行的蒙特卡洛模拟的结果表明,加速因子为45,检测次数损失适中,异常检测次数显着减少。带有真实记录的实验结果表明,其性能与模拟结果一致。

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