Dual-Microphone Gain Processing for Low-Delay Stationary Noise Suppression
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Solution Overview
Problem
Conventional sound processing methods, such as spectral subtraction, delay the signal processing of stationary noise suppression, leading to reduced call quality in hands-free devices due to the need for converting input signals into frequency domain and back, causing significant processing delays.
Innovation Solution
A sound processing apparatus that calculates gains based on signals from multiple microphones and applies these gains to reduce the amplitude of stationary noise signals, allowing for real-time processing on the time axis without converting the signal to the frequency domain, thereby minimizing delay.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If spectral subtraction method is used to suppress stationary noise, then noise suppression performance is improved, but processing delay increases due to frequency domain conversion
Solution Approach 1:
The patent replaces the frequency domain processing mechanism (Fourier transform-based spectral subtraction) with a time domain processing mechanism (autocorrelation-based gain calculation). This substitution eliminates the need for complex frequency domain conversions while achieving effective stationary noise suppression through time domain signal analysis and gain application.
2Reliability
If synchronous subtraction method is used to suppress sound from specific direction, then directional sound suppression is improved, but suppression of sounds from multiple directions deteriorates
Solution Approach 1:
The patent changes the processing parameter from simple synchronous subtraction to autocorrelation-based gain calculation. By analyzing the temporal correlation structure of the signal and applying frequency-dependent gains in the time domain, the system adaptsively suppresses stationary noise from multiple directions while preserving speech signals, overcoming the limitation of fixed directional suppression.
Data Source
AI summary
A sound processing apparatus includes a first calculator that calculates first power based on a first signal received by a first microphone that is among the first microphone and a second microphone; a second calculator that calculates second power based on a second signal received by the second microphone; a gain calculator that calculates a gain on the basis of the ratio of the first power to the second power; and a multiplier that processes the second signal using the gain calculated by the gain calculator.