Dynamic Speech Equalization for Loudspeaker Intelligibility
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Existing loudspeaker systems, particularly in power-constrained environments, suffer from suboptimal frequency content in incoming speech signals due to factors like poor microphone design, noisy surroundings, and impaired communication channels, leading to reduced listening comfort and speech intelligibility.
Innovation Solution
A method for dynamic speech equalization that dynamically applies a compensation filter and a transparent filter to incoming audio signals based on frequency content, using speech analysis to enhance frequency response and improve intelligibility.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If dynamic speech equalization with multiple compensation filters is applied, then speech intelligibility and frequency response are improved, but device complexity and computational requirements increase
Solution Approach 1:
The system dynamically adjusts filter parameters and selection based on real-time speech signal analysis. The equalization function switches between different compensation filters (first compensation filter for impaired signals, second compensation filter for less impaired signals) depending on the analyzed speech characteristics, making the system adaptive rather than static.
Solution Approach 2:
The frequency spectrum is divided into multiple bands (low frequency band below 2nd critical band, mid frequency between 2nd-4th critical bands, high frequency above 4th critical band) with different compensation strategies applied to each. This segmentation allows targeted processing of specific frequency regions affected by different types of impairments.
2Reliability
If real-time frequency analysis and dynamic filter application are implemented, then listening comfort is improved, but processing time and computational load increase
Solution Approach 1:
The system performs preliminary analysis of the speech signal to detect impairments and pre-selects appropriate compensation filters before full equalization processing. The speech signal is analyzed for characteristics indicating impairment type and severity, and the suitable filter is prepared in advance, reducing overall processing time.
Solution Approach 2:
The system applies full dynamic equalization processing only when speech impairment is detected. For clear speech signals or non-speech content, the system uses a simplified processing path or bypasses equalization entirely, reducing computational load during periods when full processing is not needed.
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.