Adaptive Beamforming Microphone Array for Noise Rejection
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Solution Overview
Problem
In noisy environments, existing communication devices struggle to effectively separate the user's voice from background noise, leading to reduced signal-to-noise ratios and increased listening fatigue.
Innovation Solution
A method and system that combines two audio signals by phase and amplitude matching the target signal portions, using a bass-boost filter and adaptive filtering to minimize noise contributions, and regulating the directivity pattern to direct noise away from the target sound source.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Object-affected harmful factors
If directional microphones with fixed directivity patterns are used to minimize noise, then noise rejection is improved, but adaptability to different noise environments and positions is worsened
Solution Approach 1:
The patent implements adaptive filtering that dynamically adjusts the directivity pattern in real-time based on the acoustic environment. The system continuously processes audio signals from multiple microphones and adapts the beamforming weights to optimize noise rejection for different noise positions and characteristics, transforming the fixed directivity pattern into a dynamic, adaptable system.
Solution Approach 2:
The system uses feedback from the audio signals processed by the adaptive filter to continuously monitor the noise environment and adjust the directivity pattern accordingly. The adaptive filtering algorithm analyzes the incoming signals and modifies the beamforming parameters in real-time, creating a closed-loop system that responds to changing acoustic conditions.
2Object-affected harmful factors
If multiple microphones are used to improve signal-to-noise ratio, then noise attenuation is improved, but device complexity is worsened
Solution Approach 1:
The patent combines multiple microphone signals through adaptive beamforming and merging them into a single enhanced output signal. The system processes signals from multiple microphones simultaneously, combining their information through weighted summation controlled by the adaptive filter, thereby improving signal-to-noise ratio while managing complexity through integrated signal processing.
3Measurement precision
If phase and amplitude matching is performed on target signals, then target signal enhancement is improved, but processing complexity is worsened
Solution Approach 1:
The adaptive filter modifies the parameters (phase and amplitude) of the target signal components in the audio signals processed by the beamformer. By dynamically adjusting these parameters based on the acoustic environment and target position, the system achieves precise matching of target signals while the adaptive nature of the filtering reduces the need for manual parameter tuning.
Applied Scientific Principles
This section explains which scientific principles are used to turn an abstract innovation direction into a practical engineering solution.
Function Achieved in This Case
This approach attenuates background noise by 3-12 dB or more, improving the voice-to-noise ratio and reducing listening fatigue by ensuring precise matching and constructive interference of target signals while averaging out noise signals.
Implementation Method 1
it is ensured that the target signal portions are cancelled out and not carried on to the subtraction output of step d). Thus only the contribution from the noise portions (or the unintended parts) of the audio signals to the system output is minimised during the processing of the subtraction output in step f). Further, it is ensured that the target portions appear maximally in the summation output from step e) due to constructive interference
Implementation Method 2
A method and system that combines two audio signals by phase and amplitude matching the target signal portions, using a bass-boost filter and adaptive filtering to minimize noise contributions, and regulating the directivity pattern to direct noise away from the target sound source.
Data Source
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AI summary
A method of combining at least two audio signals for generating an enhanced system output signal is described. The method comprises the steps of: a) measuring a sound signal at a first spatial position using a first transducer, such as a first microphone, in order to generate a first audio signal comprising a first target signal portion and a first noise signal portion, b) measuring the sound signal at a second spatial position using a second transducer, such as a second microphone, in order to generate a second audio signal comprising a second target signal portion and a second noise signal portion, c) processing the first audio signal in order to phase match and amplitude match the first target signal with the second target signal within a predetermined frequency range and generating a first processed output, d) calculating the difference between the second audio signal and the first processed output in order to generate a subtraction output, e) calculating the sum of the second audio signal and the first processed output in order to generate a summation output, f) processing the subtraction output in order to minimise a contribution from the noise signal portions to the system output signal and generating a second processed output, and g) calculating the difference between the summation output and the second processed output in order to generate the system output signal.