Non-Uniform Linear Microphone Array for Uniform Sound Gain
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Solution Overview
Problem
Conventional linear microphone arrays with equally spaced microphones struggle to achieve balanced sound collection across different frequencies, leading to inconsistent sound quality and increased ambient noise, especially when the user moves.
Innovation Solution
A microphone apparatus with a linear array of microphones arranged in a non-uniform configuration, coupled with an integrated circuit that processes merged sound signals using time-domain filters to generate output sound signals with uniform gain across various frequencies and angles.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Ease of manufacture
If microphones are arranged at equal intervals in a linear array, then the structure is simple and easy to manufacture, but the sound gain varies significantly at different sound-receiving angles and frequencies
Solution Approach 1:
The patent applies asymmetry by arranging microphones at non-uniform intervals along the linear array, specifically with different spacing distances between adjacent microphones. This asymmetric configuration creates a beamforming pattern that maintains more uniform sound gain across different sound-receiving angles and frequencies, resolving the contradiction between structural simplicity and sound gain uniformity.
2Area of stationary object
If the microphone distance is increased to widen the sound range, then more ambient sound is captured, but the sound quality deteriorates and noise increases
Solution Approach 1:
The patent applies local quality by assigning different spacing intervals to different sections of the microphone array. Specifically, microphones are positioned with varying distances from each other, where closer microphones capture direct sound better while farther microphones capture ambient sound. The signal processing unit then selectively combines these signals based on the sound-receiving angle and frequency, achieving both wide sound range and high sound quality by optimizing the contribution of each local microphone group.
3Area of stationary object
If the frequency of the sound signal is increased, then the sound range is narrowed, but the change in sound field shape becomes more severe
Solution Approach 1:
The patent applies dynamics by implementing adaptive signal processing that dynamically adjusts the beamforming weights and signal combination strategy based on the detected sound-receiving angle and frequency. The signal processing unit continuously optimizes the contribution of each microphone according to real-time acoustic conditions, allowing the system to maintain stable sound field characteristics across varying frequencies while preserving an wide sound range.
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
The solution ensures consistent sound gain at each sound-receiving angle for different frequencies, improving sound quality and reducing ambient noise, even when the user's position changes.
Implementation Method 1
The integrated circuit is configured to process a merged sound signal from different combinations of microphones using a time-domain filter to generate an output sound signal
Data Source
AI summary
A microphone apparatus is provided. The microphone apparatus includes a microphone array and an integrated circuit. The microphone array includes at least three microphones arranged in a straight line with a non-uniform configuration. The integrated circuit is electrically connected to the microphone array. The integrated circuit is configured to process a merged sound signal from different combinations of microphones using a time-domain filter to generate an output sound signal. The sound gain of the output sound signal at each sound-receiving angle of each frequency is substantially uniform.