Automatic DSP Tuning for Room-Adaptive Audio Conferencing
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Existing audio conferencing systems require significant manual effort and time for optimal configuration and maintenance, as they are sensitive to changes in room layout and audio device placement, leading to sub-optimal performance over time.
Innovation Solution
An audio system that automatically identifies devices and adjusts digital signal processing configurations based on room characteristics and device information, using a processor to send test signals, capture responses, and adjust DSP settings accordingly.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If manual configuration and testing of audio system settings is performed, then optimal audio performance can be achieved, but significant time and labor effort are required
Solution Approach 1:
The system performs self-configuration and self-optimization by automatically generating test signals, capturing microphone responses, analyzing the audio environment, and adjusting DSP parameters without human intervention. The audio system configures itself based on the measured room acoustics and device characteristics.
Solution Approach 2:
The system performs preliminary measurements by playing test signals through speakers and capturing microphone responses before final system deployment. This advance characterization of the audio environment enables optimized configuration settings to be established prior to actual use.
2Adaptability or versatility
If manual configuration is performed to adapt to room layout changes, then audio system performance can be optimized, but frequent manual intervention is required
Solution Approach 1:
The system continuously monitors the audio environment by capturing microphone responses to test signals and uses this feedback to automatically detect changes in room layout or device positioning. Based on this feedback, the system re-optimizes DSP parameters without requiring manual reconfiguration.
Solution Approach 2:
The system transitions from static manual configuration to dynamic automatic adaptation. It continuously measures the audio environment and adjusts configuration parameters in real-time based on detected changes, making the system adaptive to evolving conditions without human intervention.
3Reliability
If comprehensive testing of audio signals and continuity verification is performed, then system reliability is improved, but setup complexity increases
Solution Approach 1:
The system replaces manual mechanical testing procedures with automated electronic signal generation and analysis. Test signals are automatically played through speakers, captured by microphones, and analyzed by processing circuits to verify signal continuity and quality without requiring physical manual testing.
Solution Approach 2:
The same audio processing circuits and components used for normal audio operation are utilized for testing and verification purposes. The system performs multiple functions including signal generation, capture, analysis, and optimization using a unified automated process rather than separate dedicated testing equipment.
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
Facilitates rapid, efficient setup and maintenance of audio conferencing systems by optimizing settings for acoustic echo cancellation, noise reduction, and frequency shaping, reducing manual intervention and ensuring consistent performance.
Implementation Method 1
a speaker configured to play an audio test signal; at least one microphone configured to capture an audio response signal representative of the audio test signal
Data Source
AI summary
Embodiments include a processing device communicatively coupled to a plurality of audio devices comprising at least one microphone and at least one speaker, and to a digital signal processing (DSP) component having a plurality of audio input channels for receiving audio signals captured by the at least one microphone, the processing device being configured to identify one or more of the audio devices based on a unique identifier associated with each of the one or more audio devices; obtain device information from each identified audio device; and adjust one or more settings of the DSP component based on the device information. A computer-implemented method of automatically configuring an audio conferencing system, comprising a digital signal processing (DSP) component and a plurality of audio devices including at least one speaker and at least one microphone, is also provided.