Automatic Speaker Equalization Using Room Position Feedback
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Solution Overview
Problem
Manual audio equalization is cumbersome and requires advanced knowledge, making it difficult for average users to achieve high-quality audio output in various environments without repeated setup and calibration processes.
Innovation Solution
Implementing automatic audio equalization using microphones integrated into electronic devices, which measure phase shifts and relative amplitude spectral features to adjust frequency responses without user input, utilizing machine learning and acoustical models to learn and recognize patterns for optimal audio correction.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Manufacturing precision
If manual audio equalization is performed, then audio output quality can be improved, but the process becomes cumbersome and time-consuming requiring user knowledge
Solution Approach 1:
The system performs automatic audio equalization using the device's own microphones to capture and analyze its speaker output in the actual listening environment. The processor automatically generates and applies equalization filters based on the measured frequency response, enabling the device to self-calibrate without user intervention or external equipment.
Solution Approach 2:
The patent replaces the manual mechanical process of equalization (requiring physical microphones, recording devices, and user manipulation) with an automated electronic system. The device uses its integrated microphones and processor to automatically measure, analyze, and correct frequency response issues through software-based equalization filters.
2Measurement precision
If manual equalization is performed with external microphones, then frequency response can be measured, but the process requires advanced knowledge and repeated calibration when room changes
Solution Approach 1:
The system uses the device's own integrated microphones to perform self-measurement of its speaker output in the actual listening environment. This eliminates the need for external measurement equipment and repeated calibration procedures, as the device automatically adapts to its specific placement and room acoustics each time it is used.
Solution Approach 2:
The integrated microphones serve dual functions: they act as both the sound source receiver for equalization measurement and the functional microphone for the device's primary audio processing tasks. This multi-functionality eliminates the need for separate calibration equipment and reduces the overall complexity of the equalization process.
3Power
If speakers are placed near boundaries to enhance bass, then low frequency output is improved, but echo and boomy bass problems occur
Solution Approach 1:
The system uses the device's microphones to capture the actual acoustic output in the listening environment, creating a feedback loop that measures the frequency response including any boomy bass or echo issues. The processor then generates equalization filters that apply corrective attenuation to problematic frequency ranges, automatically compensating for the harmful effects of boundary placement.
Solution Approach 2:
The system dynamically changes the electrical parameters of the speaker output through software-based equalization filters. By adjusting the gain and frequency response characteristics in real-time based on measured conditions, the system can compensate for excessive bass boost and echo problems caused by placement near boundaries or reflective surfaces.
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
Enables high-quality audio output in any environment without user interaction, simplifying the process and eliminating the need for manual calibration, ensuring consistent sound quality regardless of device placement or room acoustics.
Implementation Method 1
obtaining a collection of audio content signals by receiving the outputted audio content at each microphone of the plurality of microphones
Implementation Method 2
measuring a plurality of phase differences between the outputted audio content and the received audio content at the microphones
Data Source
AI summary
The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.