Hearing System Adaptive Beamformer Distance Estimation
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Solution Overview
Problem
Body-worn hearing systems face challenges in effectively picking up a user's voice in communication modes due to variations in sound pressure levels between microphones caused by differences in distance from the mouth, especially in near-field applications, where traditional directional microphone systems fail to account for both direction and distance in optimizing directional response.
Innovation Solution
The system estimates the distance and relative transfer functions between the user's mouth and microphones, and updates beamformer weights based on these parameters to adaptively filter and enhance the user's voice, using a dictionary of pre-stored values for different geometric configurations, including tilt angles, to improve noise reduction and directional performance.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If traditional directional microphone systems are used, then the system structure is simple, but the directional response accuracy deteriorates in near-field applications because they do not account for both direction and distance
Solution Approach 1:
The system dynamically adapts beamformer weights based on estimated distance and direction parameters. The beamforming configuration changes in real-time according to the user's mouth position relative to the microphones, transitioning from static to dynamic directional response optimization for near-field conditions
Solution Approach 2:
The system changes key parameters including beamformer weights, distance estimates, and direction vectors to optimize directional response. By adjusting these parameters based on near-field acoustic characteristics, the system achieves accurate directional filtering that accounts for both direction and distance
2Reliability
If the microphone unit is placed close to the mouth for better voice pickup, then the voice signal strength increases, but the sound pressure level difference between microphones increases making directional filtering more difficult
Solution Approach 1:
The system uses feedback from distance estimation and acoustic signal analysis to continuously adjust beamformer weights. By monitoring the acoustic environment and mouth position, the system adapts its filtering characteristics to maintain optimal voice pickup quality despite varying microphone-to-mouth distances
Solution Approach 2:
The system performs preliminary distance estimation and beamformer weight calculation before actual voice transmission. By pre-computing the appropriate beamforming configuration based on estimated geometry, the system prepares the optimal filtering state in advance, reducing the complexity of real-time adaptation
3Measurement precision
If beamformer weights are updated continuously for optimal performance, then the voice pickup accuracy improves, but the computational load and processing time increase
Solution Approach 1:
The system updates beamformer weights periodically based on detected changes in mouth position or acoustic environment rather than continuously. This periodic update approach maintains voice pickup accuracy while reducing computational load and processing time by updating only when necessary
Solution Approach 2:
The system applies partial updates to beamformer weights, adjusting only the parameters that have changed significantly rather than re-computing all weights. This selective update strategy maintains accuracy for the most critical parameters while reducing overall processing requirements
Data Source
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Figure 3
AI summary
A body worn hearing system comprises a hearing device, e.g. a hearing aid, and a separate microphone unit for picking up a voice of the user. The hearing device comprises a forward path comprising an input unit for providing an electric input signal representative of sound in the environment, a signal processing unit for providing a processed signal, and an output unit for generating stimuli perceivable as sound when presented to the user based on said processed signal. The microphone unit comprises a multitude M of microphones, and a multi-input noise reduction system for providing an estimate Ŝ of a target signal s comprising the user's voice, and comprising a multi-input beamformer filtering unit operationally coupled to said multitude of microphones. The hearing device and the microphone unit are configured to receive and transmit an audio signal from/to a communication device, respectively, and for establishing a communication link between them for exchanging information. The hearing system comprises a control unit configured to estimate a current distance between the user's mouth and the microphone unit, and to control the multi-input noise reduction system in dependence of said distance.