Microphone Array Direction Estimation via FFT Phase Analysis
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Solution Overview
Problem
Existing sound collecting devices face limitations in accurately estimating the direction of arrival of a sound source due to the inability to detect time differences of arrival with precision less than one sample, which is influenced by the sampling frequency and microphone spacing.
Innovation Solution
A sound collecting device with a signal processing unit that calculates the total time difference of arrival by adding a first difference in time, based on inter-sample accuracy, and a second difference, to estimate the sound source position, using a combination of echo cancellation, voice activity detection, and beam forming processes to enhance accuracy.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If cross-correlation function is used to find time difference of arrival in time domain, then the method is simple and widely applicable, but the measurement precision is limited to one sample or less
Solution Approach 1:
The patent transforms the time domain signal processing into the frequency domain by applying Fast Fourier Transform (FFT). This dimensional change allows sub-sample precision in time difference estimation through phase difference calculation, overcoming the one-sample limitation of time domain cross-correlation methods.
Solution Approach 2:
The patent replaces the mechanical/time-domain cross-correlation approach with a frequency-domain spectral analysis approach. By using autocorrelation in the frequency domain and calculating phase differences, the system achieves higher precision without requiring complex time-domain processing.
2Measurement precision
If sampling frequency and microphone spacing are increased to improve measurement precision, then the direction of arrival estimation accuracy improves, but the device complexity and cost increase
Solution Approach 1:
The patent changes the parameter domain from time to frequency by using FFT. This allows the system to achieve sub-sample precision without increasing the physical spacing between microphones or the sampling frequency, as the phase information in the frequency domain provides finer resolution.
Solution Approach 2:
The patent introduces an intermediary processing layer (FFT and spectral analysis) between the microphone array and the direction estimation. This intermediary layer extracts phase difference information that enables high-precision measurement without requiring closer microphone spacing or higher sampling rates.
3Measurement precision
If more microphones are added to the array to improve measurement precision, then the direction of arrival estimation accuracy improves, but the device complexity increases
Solution Approach 1:
The patent segments the signal processing into independent modules: FFT transformation, autocorrelation calculation, phase difference computation, and direction estimation. This modular approach allows the system to handle multiple microphones efficiently by processing their signals through standardized stages, reducing overall system complexity.
Solution Approach 2:
The patent creates a universal processing framework using FFT and spectral analysis that can handle variable numbers of microphones in the array. The same autocorrelation and phase difference calculation methodology applies regardless of the specific microphone configuration, making the system adaptable without increasing complexity.
Applied Scientific Principles
This section explains which scientific principles are used to turn an abstract innovation direction into a practical engineering solution.
Function Achieved in This Case
This approach improves the accuracy of sound source direction estimation, allowing for more precise localization of sound sources, even when time differences are less than one sample, by incorporating echo cancellation and beam forming techniques.
Implementation Method 1
a Fast Fourier Transform is applied to the collected sound signals of the plurality of microphones, and a phase difference between the collected sound signals is calculated
Implementation Method 2
The difference in time of arrival is converted to a distance by multiplying by the speed of sound
Data Source
AI summary
A signal processing unit calculates a first difference in time of arrival of sound from a sound source to a first and to a second microphone comprising a microphone array and calculates a second difference in time of arrival, which is the difference between the first difference in time of arrival and an actual time, of arrival, and determines the position of the sound source based on the sum of the first difference in time of arrival and the second difference in time of arrival.