Microphone Array Beamforming for Moving Interference Suppression
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Existing speech enhancement technologies using generalized sidelobe cancellers (GSC) fail to account for the movement of interfering sound sources, leading to poor sound source separation and increased interference leaks.
Innovation Solution
An audio signal processing method that updates filters in real-time based on the steering vector changes of a target sound source, using a combination of first, second, and third filters to adaptively track and suppress interference, enhancing target speech accuracy.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If a pre-designed filter is used in GSC algorithm, then the beam performance is improved, but the sound source separation effect deteriorates when interference moves
Solution Approach 1:
The patent applies dynamics by transitioning from a static pre-designed filter to a dynamic adaptive filter that continuously updates its parameters based on real-time environmental conditions. The filter adapts to moving interference sources by adjusting its weight vectors and steering vectors dynamically, thereby maintaining both beam performance and sound source separation effectiveness in changing acoustic environments.
Solution Approach 2:
The patent implements feedback mechanisms where the filter continuously monitors the acoustic environment and adjusts its parameters based on feedback from the actual interference patterns. The adaptive filter uses the observed interference characteristics to update its weight vectors, creating a closed-loop system that maintains optimal performance despite source movement.
2Adaptability or versatility
If filters are updated in real-time based on steering vector changes, then the tracking performance is improved, but the device complexity increases
Solution Approach 1:
The patent applies self-service by designing an adaptive filter system that automatically updates its own parameters without external intervention. The filter monitors the acoustic environment itself and performs real-time parameter adjustments based on detected steering vector changes, eliminating the need for manual recalibration or complex external control systems.
Solution Approach 2:
The patent implements parameter changes by dynamically modifying the filter's weight vectors and steering vectors in response to detected changes in the acoustic environment. This allows the filter to adapt to moving sources through continuous parameter adjustment rather than requiring complete filter redesign, thereby managing complexity while maintaining tracking performance.
Data Source
Figure 1~2(2)
Figure 3(1)~4
Figure 5~6
AI summary
The present application relates to the field of speech processing. Disclosed are an audio signal processing method, a device, equipment, and a storage medium. The method comprises: acquiring audio signals collected by different microphones in a microphone array; filtering the audio signals via a first filter to produce a first target beam; filtering the audio signals via a second filter to produce a first interference beam; acquiring a second interference beam of the first interference beam via a third filter; determining the difference between the first target beam and the second interference beam as a first audio processing output; self-adaptively updating at least one of the second filter and the third filter, and, upon completion of the update, updating the first filter on the basis of the second filter and of the third filter. The method reduces interference leakage in a case of interfered movement.