Microphone Array Direction Estimation for Immersive Audio
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Current methods for generating surround sound using microphone arrays are limited in their ability to accurately estimate sound direction and effectively process signals to produce immersive audio experiences, particularly in environments with diffuse sound sources.
Innovation Solution
The proposed solution involves analyzing signals from a microphone array to estimate time differences and direction of arrival, using a combination of filters with transfer functions that vary based on spatial orientations to compute loudspeaker driving signals, which are then processed to generate surround sound suitable for playback on multi-channel speaker systems.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If traditional stereo or mono sound channels are used, then the system complexity is low, but the sound stage and audio immersion are limited
Solution Approach 1:
The patent segments the audio signal processing into multiple independent channels (front left, front right, center, surround left, surround right, LFE) that can be processed and reproduced separately. Each channel is handled by dedicated loudspeakers positioned around the listener, allowing the system to create immersive surround sound without requiring a monolithic complex processing unit.
Solution Approach 2:
The patent transitions from traditional two-dimensional stereo sound (front and back) to three-dimensional surround sound by adding spatial depth through multiple channels positioned around the listener. This includes front channels, center channel, surround channels behind the listener, and low-frequency effects, creating a comprehensive spatial audio environment that envelops the listener in three dimensions.
2Measurement precision
If microphone arrays are used to capture sound from multiple directions, then the sound direction estimation capability is improved, but the accuracy in diffuse sound environments deteriorates
Solution Approach 1:
The patent employs dynamic filter transfer functions that adapt based on the estimated direction of arrival of sound sources. The filters are not static but are adjusted in real-time according to the spatial characteristics of the incoming sound, allowing the system to maintain accurate direction estimation even when sound sources move or when the acoustic environment changes to become more diffuse.
Solution Approach 2:
The patent changes the parameters of the filter transfer functions based on the estimated sound direction and spatial orientation. By adjusting filter parameters dynamically according to the measured acoustic conditions, the system can optimize its performance for different sound environments, including diffuse sound fields, maintaining reliability across varying conditions.
3Manufacturing precision
If filter transfer functions with spatial orientation components are used, then the surround sound quality is improved, but the signal processing complexity increases
Solution Approach 1:
The patent segments the filter processing into distinct spatial orientation components (front-back, left-right) that can be processed independently and then combined. This segmentation allows the complex three-dimensional spatial filtering to be broken down into manageable two-dimensional components, reducing the overall processing complexity while maintaining high surround sound quality.
Solution Approach 2:
The patent designs filter transfer functions that serve multiple purposes: they perform frequency filtering, spatial direction estimation, and surround channel separation simultaneously. This multi-functionality reduces the need for separate processing stages, thereby reducing overall signal processing complexity while achieving high-quality surround sound reproduction.
Data Source
AI summary
A signal from each of an array of microphones is analyzed. For at least one subset of microphone signals, a time difference is estimated, which characterizes the relative time delays between the signals in the subset. A direction is estimated from which microphone inputs arrive from one or more acoustic sources, based at least partially on the estimated time differences. The microphone signals are filtered in relation to at least one filter transfer function, related to one or more filters. A first filter transfer function component has a value related to a first spatial orientation of the arrival direction, and a second component has a value related to a spatial orientation that is substantially orthogonal in relation to the first. A third filter function may have a fixed value. A driving signal for at least two loudspeakers is computed based on the filtering.


