Ad-Hoc麦克风阵列多通道记录的无源时间偏移估计

Pasi Pertilä, M. Hämäläinen, Mikael Mieskolainen
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引用次数: 36

摘要

近年来,自组织麦克风阵列已经变得无处不在,捕获硬件和质量也越来越复杂。Ad-hoc阵列在音频应用中具有巨大的潜力,但它们本质上是异步的,即每个通道中存在时间偏移,而且设备位置通常是未知的。因此,这些数据并不直接适用于传统的麦克风阵列应用,如源定位和波束形成。本文提出了一种用于静态ad-hoc麦克风阵列时间偏移估计的最小二乘方法。该方法利用所捕获的音频内容,而不需要发射校准信号,只要在记录期间有足够数量的声源环绕所述阵列。给出了估计量的Cramer-Rao下界,并研究了有限数量的周围源对解精度的影响。然后给出了采用非线性滤波和自动参数调整的实际实现。在混响和噪声水平范围内的仿真证明了该算法的鲁棒性。使用智能手机,当算法的假设得到满足时,平均均方根误差为3.5个样本(在48 kHz时)。
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Passive Temporal Offset Estimation of Multichannel Recordings of an Ad-Hoc Microphone Array
In recent years ad-hoc microphone arrays have become ubiquitous, and the capture hardware and quality is increasingly more sophisticated. Ad-hoc arrays hold a vast potential for audio applications, but they are inherently asynchronous, i.e., temporal offset exists in each channel, and furthermore the device locations are generally unknown. Therefore, the data is not directly suitable for traditional microphone array applications such as source localization and beamforming. This work presents a least squares method for temporal offset estimation of a static ad-hoc microphone array. The method utilizes the captured audio content without the need to emit calibration signals, provided that during the recording a sufficient amount of sound sources surround the array. The Cramer-Rao lower bound of the estimator is given and the effect of limited number of surrounding sources on the solution accuracy is investigated. A practical implementation is then presented using non-linear filtering with automatic parameter adjustment. Simulations over a range of reverberation and noise levels demonstrate the algorithm's robustness. Using smartphones an average RMS error of 3.5 samples (at 48 kHz) was reached when the algorithm's assumptions were met.
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来源期刊
IEEE Transactions on Audio Speech and Language Processing
IEEE Transactions on Audio Speech and Language Processing 工程技术-工程:电子与电气
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0.00%
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0
审稿时长
24.0 months
期刊介绍: The IEEE Transactions on Audio, Speech and Language Processing covers the sciences, technologies and applications relating to the analysis, coding, enhancement, recognition and synthesis of audio, music, speech and language. In particular, audio processing also covers auditory modeling, acoustic modeling and source separation. Speech processing also covers speech production and perception, adaptation, lexical modeling and speaker recognition. Language processing also covers spoken language understanding, translation, summarization, mining, general language modeling, as well as spoken dialog systems.
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