Loudspeaker Speech Equalization Using Adaptive Compensation Filters
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Solution Overview
Problem
Existing loudspeaker systems, particularly in applications like speakerphones and telecommunication devices, often suffer from suboptimal frequency content in incoming speech signals due to poor microphone design, noise, and channel limitations, leading to reduced listening comfort and speech intelligibility.
Innovation Solution
A method for dynamic speech equalization that involves analyzing incoming audio signals using a speech analysis block to apply frequency-dependent adjustments through a combination of compensation and transparent filters, enhancing the signal's frequency response to match ideal speech characteristics.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If dynamic speech equalization with multiple filters is applied, then speech intelligibility is improved, but device complexity increases
Solution Approach 1:
The equalization function is segmented into multiple independent compensation filters (first compensation filter, second compensation filter, etc.), each targeting specific frequency ranges or impairment types. This allows the system to process different frequency components separately and combine them to achieve comprehensive speech enhancement while maintaining modular architecture that manages complexity.
Solution Approach 2:
The system dynamically selects and adjusts which compensation filters to apply based on real-time analysis of the input signal characteristics. The equalization unit can adaptively activate or deactivate specific filters depending on the detected speech conditions, making the processing complexity variable rather than fixed, thus balancing performance with computational load.
2Measurement precision
If frequency-dependent processing is applied to correct suboptimal frequency content, then listening comfort is improved, but processing time increases
Solution Approach 1:
The system performs preliminary analysis of the input signal to identify speech components and their frequency characteristics before applying the full equalization processing. This preliminary detection allows the system to prepare appropriate filter configurations in advance, reducing the actual processing time when enhancement is needed while maintaining listening comfort quality.
3Measurement precision
If multiple compensation filters are dynamically applied, then speech intelligibility is improved, but computational power consumption increases
Solution Approach 1:
The equalization unit dynamically activates only the necessary compensation filters based on real-time speech analysis, rather than continuously applying all filters. This adaptive approach ensures that computational resources and power are consumed only when and where needed to improve speech intelligibility, optimizing the balance between performance and energy efficiency.
Data Source
AI summary
A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.