Adaptive Microphone Array With User Feedback
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Solution Overview
Problem
Current methods for reducing noise in conference calls and similar communications are unsatisfactory, particularly with non-stationary noises, and existing solutions like beamforming do not effectively suppress reverberation or multiple voices, while conventional digital signal processing algorithms fail to adapt to everyday noise variability, leading to suboptimal sound quality and user discomfort.
Innovation Solution
The use of adaptive microphone arrays integrated with machine learning models that incorporate user feedback to optimize noise suppression, allowing for real-time noise cancellation and speech enhancement by generating anti-noise signals based on sound source localization and user satisfaction inputs, enabling improved sound quality for both local and remote users.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If conventional digital signal processing algorithms are used to suppress noise, then stationary noise can be reduced, but the system cannot adapt to non-stationary noise and everyday noise variability
Solution Approach 1:
The patent implements dynamic adaptability by enabling the microphone array system to automatically adjust its noise suppression parameters and beamforming patterns in real-time based on changing acoustic environments. The system transitions from static conventional DSP algorithms to dynamic adaptive processing that responds to non-stationary noise conditions, speech presence detection, and environmental variations, thereby resolving the contradiction between reliable noise suppression and adaptability to noise variability.
Solution Approach 2:
The system incorporates feedback mechanisms where the output of noise suppression and beamforming processes is continuously monitored and fed back to adjust processing parameters. This feedback loop enables the system to learn from its performance and adapt to different noise scenarios, improving both reliability in noise suppression and versatility in handling various noise types including non-stationary and reverberant environments.
2Measurement precision
If beamforming techniques are used to isolate speech, then speech isolation can be achieved, but the system degrades when multiple voices are present and cannot suppress reverberation or noise from the same direction
Solution Approach 1:
The patent enhances beamforming universality by integrating multiple processing functions into a unified microphone array system. The system performs speech isolation, multi-voice separation, reverberation suppression, and noise cancellation simultaneously through coordinated beamforming patterns and post-processing techniques. This multi-functional approach allows the system to maintain speech isolation capability while adapting to diverse scenarios including multiple speakers, reverberant rooms, and various noise conditions.
Solution Approach 2:
The system employs composite signal processing techniques that combine beamforming outputs with noise suppression filters and reverberation reduction algorithms. By layering multiple processing stages and combining their effects, the system achieves robust speech isolation that works across diverse acoustic environments, effectively handling cases where single techniques would fail.
3Reliability
If headphones are used to improve sound quality, then audio quality can be enhanced, but user comfort is reduced and user mobility is impeded
Solution Approach 1:
The patent extracts the noise suppression and speech enhancement functions from the user's ears (where headphones would be required) and implements them in the acoustic environment using microphone arrays and signal processing. By taking out the active noise control function from the headphones and placing it in the environment, the system achieves high sound quality without requiring the user to wear headphones, thereby maintaining comfort and mobility.
Data Source
AI summary
One example method includes performing sound quality operations. Microphone arrays are used to cancel or reduce or suppress background noise and to enhance speech. Subjective user input is received by an orchestration engine. The orchestration engine generates an output that includes at least adjustments to a microphone array. Controlling the microphone array based, in part, on subjective user feedback, allows desired speech or desired sound to be heard more clearly by the user.