Audio Beamforming Using Eigenbeams for Directivity and Noise Cancellation
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Solution Overview
Problem
Conventional audio beamforming algorithms, such as additive delay-and-sum, fail to provide sufficient directivity when the wavelength of sound is larger than the microphone array, leading to poor performance in directional sensitivity and increased sensitivity to noise and microphone mismatches.
Innovation Solution
An audio beamforming approach using non-adaptive reference beams, specifically Eigenbeams or orthogonal beams, generated by a constant combination of signals from multiple microphones, which allows for efficient cancellation of single point interference and diffuse noise, with adaptive beam steering and sidelobe cancellation techniques to maximize directivity and minimize noise impact.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If superdirective beamforming techniques are applied to improve directivity, then directivity is improved, but sensitivity to white noise and microphone mismatches increases
Solution Approach 1:
The patent changes the fundamental parameters of the beamforming approach by using additive delay-and-sum beamforming with carefully designed time delays and phase shifts, rather than superdirective techniques. This parameter change allows achieving good directivity without the harmful sensitivity to noise and mismatches that characterizes superdirective beamforming.
Solution Approach 2:
The patent employs simple, robust delay-and-sum operations that are computationally inexpensive and numerically stable, rather than complex superdirective filters. This approach uses simpler processing that is less sensitive to implementation errors and environmental variations, effectively trading some theoretical directivity performance for robustness and reliability.
2Ease of manufacture
If additive delay-and-sum beamforming is used, then implementation is simple, but directivity deteriorates when wavelength is much larger than array size
Solution Approach 1:
The patent introduces dynamic adaptability to the simple delay-and-sum framework by continuously estimating interference directions and updating beamforming weights in real-time. This dynamic adaptation allows the system to maintain good directivity performance even when the array is small compared to the wavelength, overcoming the static limitations of conventional delay-and-sum beamforming.
Solution Approach 2:
The patent implements feedback mechanisms where the output of the beamformer is used to estimate interference directions, which then feed back into updating the beamforming weights. This feedback loop enables the system to adaptively improve its directivity performance based on actual environmental conditions, compensating for the inherent limitations of small arrays.
3Object-affected harmful factors
If complex beamforming algorithms are applied to maximize interference cancellation, then interference reduction is improved, but processing complexity and resource demand increase
Solution Approach 1:
The patent segments the beamforming process into distinct functional blocks: interference direction estimation, weight calculation, and signal combination. Each block performs a specific, relatively simple function, making the overall complex task manageable and efficient to implement. This segmentation allows using straightforward algorithms like power spectral density estimation and closed-form weight solutions rather than requiring complex iterative optimization.
Solution Approach 2:
The patent replaces complex mechanical or computational beamforming systems with an electronic signal processing approach using simple delay-and-sum operations combined with adaptive weighting. This substitution uses efficient digital signal processing techniques that require minimal computational resources compared to more complex beamforming algorithms, achieving good interference cancellation with low processing complexity.
Data Source
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AI summary
An audio beamforming apparatus comprises a receiving circuit (103) which receives signals from an at least two-dimensional microphone array (101). A reference circuit (105) generates reference beams and a combining circuit (107) generates an output signal corresponding to a desired beam pattern by combining the reference beams. An estimation circuit (109) generates a direction estimate by determining angles corresponding to local minima for a power measure of the output signal in at least a first and respectively second angle interval. The direction estimate is generated by selecting one of the angles. The combining circuit (107) determines combination parameters to provide a notch in an angle corresponding to the direction estimate and a minimization of a directivity cost measure where the directivity cost measure is indicative of a ratio between a gain in the first direction and an energy averaged gain.