Microphone Array Noise Reduction via Phase Difference Estimation
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Solution Overview
Problem
Conventional microphone systems, especially directional microphone systems, face challenges in effectively reducing noise without speech distortion due to insufficient directionality and high hardware costs, particularly in indoor or in-vehicle environments.
Innovation Solution
A microphone array structure comprising at least two microphones, FFT modules, a processing module, a phase difference estimation module, a mask estimation module, and an IFFT-OLA module, which uses phase difference estimation and Golden Section Search algorithms to determine optimal ITD thresholds and apply noise reduction algorithms based on the angle between speech and noise signals, enhancing noise reduction and speech quality.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Object-affected harmful factors
If a directional microphone system is used to reduce background noise, then noise sensitivity is reduced, but speech sensitivity becomes more sensitive and directionality is insufficient
Solution Approach 1:
The patent divides the noise reduction process into multiple stages: beamforming for directional filtering, spectral subtraction for noise estimation, and phase difference estimation for speech enhancement. Each stage targets specific aspects of signal separation, allowing the system to achieve both noise reduction and speech clarity without excessive directionality constraints
Solution Approach 2:
The patent dynamically adjusts processing parameters based on signal characteristics. The phase difference estimation module calculates ITD (inter-aural time difference) and uses this to adaptively select noise reduction algorithms. The system changes processing intensity and method based on the calculated angle between speech and noise signals, optimizing performance across different spatial configurations
2Object-affected harmful factors
If numerous microphones and filters are used to reduce indoor or in-vehicle noises, then noise reduction effectiveness is improved, but hardware cost greatly increases
Solution Approach 1:
The patent replaces physical noise reduction mechanisms (additional microphones and analog filters) with digital signal processing algorithms. The processing module implements noise reduction through mathematical operations including spectral subtraction, phase difference estimation, and adaptive filtering, achieving effective noise reduction with minimal hardware additions
Solution Approach 2:
The processing module serves multiple functions: it performs beamforming for directional filtering, spectral analysis for noise estimation, phase difference calculation for speech enhancement, and adaptive algorithm selection. This multi-functional approach eliminates the need for separate dedicated hardware components for each function, reducing overall system complexity
3Object-affected harmful factors
If conventional noise reduction algorithms are applied, then noise is reduced, but speech distortion occurs due to insufficient directionality
Solution Approach 1:
The patent implements feedback through the phase difference estimation module, which continuously calculates the angle between speech and noise signals based on the processed audio. This feedback information is used to adaptively select and adjust noise reduction algorithms in real-time, allowing the system to learn from previous processing results and optimize speech preservation while maintaining noise reduction effectiveness
Solution Approach 2:
The patent applies different noise reduction strategies to different frequency components and spatial regions. The spectral subtraction module processes specific frequency bands differently based on the calculated signal angles, and the phase difference estimation applies targeted corrections to specific time-frequency components, preserving speech characteristics while removing noise
Data Source
AI summary
The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.


