Array Microphone Lobe Gain Control for Consistent Talker Levels
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Solution Overview
Problem
Existing array microphones face challenges in optimizing lobe geometry and gain settings due to inaccuracies in distance estimation, leading to inconsistent audio levels and suboptimal performance in capturing desired audio sources while rejecting unwanted noise.
Innovation Solution
The system uses sound location information to calculate a direct distance between the microphone and audio source, incorporating height information to normalize radius estimation, and adjusts lobe gain based on this improved distance calculation to ensure uniform signal levels across all input channels.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If uniform gain settings are applied to all microphone lobes, then the system is simple to operate, but audio levels become inconsistent across different talker distances
Solution Approach 1:
The system dynamically adjusts lobe gain settings based on detected talker distance. The gain for each lobe is no longer static but varies according to the distance of the audio source, allowing the system to adapt to different spatial configurations and maintain consistent audio levels across varying talker positions.
Solution Approach 2:
The patent changes the gain parameter of microphone lobes based on the distance to the audio source. By calculating distance from sound location information and adjusting gain accordingly, the system compensates for distance-related volume variations, ensuring that talkers at different distances are captured at appropriate audio levels.
2Reliability
If default lobe geometry is used for all audio sources, then the system is easy to deploy, but performance is suboptimal for different audio source scenarios
Solution Approach 1:
The system applies different lobe geometry settings to different audio sources based on their specific characteristics and locations. Rather than using a uniform default geometry for all sources, the patent tailors lobe parameters such as width and shape to the particular needs of each audio source scenario, improving capture performance for diverse situations.
Solution Approach 2:
The lobe geometry is made dynamic and adaptive rather than static. The system adjusts lobe parameters in response to detected audio source characteristics and spatial position, enabling optimal performance across varying scenarios while maintaining automatic operation without manual reconfiguration.
3Adaptability or versatility
If automatic lobe deployment is used to respond to new talkers, then the system is highly adaptive, but distance estimation inaccuracies lead to inconsistent audio levels
Solution Approach 1:
The system uses feedback from sound location information to continuously refine lobe gain settings. By monitoring the detected talker position and distance, the system adjusts gain in response to actual performance, compensating for estimation inaccuracies and maintaining consistent audio levels through continuous adaptation.
Data Source
AI summary
Systems and methods are provided for automatically adjusting lobe gain by receiving, from at least one microphone, sound location information for an audio source detected by the at least one microphone; calculating a distance between the audio source and the at least one microphone based on the sound location information, a first height associated with the at least one microphone, and a second height associated with the audio source; determining, based on the distance, a gain value for an audio pickup lobe directed towards the audio source by the at least one microphone; and applying the gain value to an audio channel of the at least one microphone, the audio channel configured to receive audio signals captured using the audio pickup lobe.