Adaptive Audio Rendering via Mobile Microphone Calibration
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Solution Overview
Problem
Conventional audio rendering systems for multichannel setups lack flexibility and user control, as they are designed for standardized loudspeaker layouts and cannot adapt to non-standard configurations or changing environments, limiting their effectiveness in dynamic settings.
Innovation Solution
A portable electronic device with multiple microphones and a user-controlled interface synchronizes with the audio rendering system via WiFi or docking station to send training signals, allowing for dynamic calibration and adaptive compensation based on real-time analysis of loudspeaker positions and room characteristics, enabling flexible and adaptive audio rendering.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If conventional audio rendering systems use standardized loudspeaker layouts, then the rendering is optimized for the standardized configuration, but the system cannot adapt to non-standard configurations or changing environments
Solution Approach 1:
The system dynamically adapts the audio rendering based on detected loudspeaker positions and room characteristics. The calibration parameters (delays, gains, equalization filters) are not fixed but are adjusted in real-time based on the detected environment, allowing the system to maintain optimized rendering across various configurations.
Solution Approach 2:
The system changes key rendering parameters such as time delays, gain levels, and equalization filters based on the detected loudspeaker layout and room acoustics. By modifying these parameters dynamically, the system achieves optimized rendering for both standardized and non-standardized configurations without requiring a fixed layout.
2Ease of operation
If automatic calibration systems are used without user control, then the calibration process is simplified, but the user loses freedom for adaptation to personal preferences
Solution Approach 1:
The calibration process is segmented into two independent stages: an automatic detection phase that identifies loudspeaker positions and room characteristics, and a user-controlled adjustment phase that allows personalization. This segmentation enables both automated simplification and user freedom to coexist.
Solution Approach 2:
The system transitions from a static, fully automatic calibration to a dynamic process where the user can intervene and adjust parameters after automatic detection. The calibration state is not fixed but can be modified by the user based on personal preferences, combining automation with adaptability.
3Measurement precision
If dedicated calibration systems with specialized microphones are used, then measurement precision is improved, but device complexity and cost increase
Solution Approach 1:
The system uses a universal mobile device (smartphone or tablet) that the user already possesses, eliminating the need for dedicated calibration equipment. The mobile device's built-in microphone is used for detection, and its processing capabilities are leveraged for analysis, achieving sufficient measurement precision without additional specialized hardware.
Solution Approach 2:
The mobile device serves multiple functions: it acts as the detection instrument, the processing unit, and the user interface. The device uses its own resources (microphone, processor, display) to perform the calibration, eliminating the need for separate dedicated calibration systems and reducing overall complexity.
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 solution allows for precise calibration and improved audio rendering in both static and dynamic environments, providing users with control over the calibration process and enhancing the accuracy of audio reproduction by considering the listener's position and room characteristics.
Implementation Method 1
a mobile device with at least two microphones, wherein the microphones are used for detecting a training signal transmitted through the loudspeakers
Data Source
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AI summary
The invention relates to an audio rendering system(100) for audio rendering room acoustics, comprising: a plurality of loudspeakers (101); a mobile device (105) comprising at least two microphones (107), the mobile device (105) being located at a room position of a user (109) using the mobile device (105); and a control device (103) coupled to the plurality of loudspeakers (101) and coupled to the mobile device (105), wherein the control device (103) is configured to calibrate the audio rendering system (100) based on the room position of the user (109) and on room positions of the loudspeakers(101) by evaluating information of the at least two microphones (107) of the mobile device (105).