Adaptive Room Equalization for Noise-Robust Timbre Control
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Solution Overview
Problem
Existing acoustic control systems face challenges in accurately estimating the room impulse response (RIR) due to external influences like background noise, which affects signal-to-noise ratio and requires significant memory to store RIRs for various fader/balance settings.
Innovation Solution
A system and method that utilize a time-to-frequency transform block, frequency-to-time transform block, loudspeaker, microphone, noise extraction block, and equalization block to automatically control the timbre of sound signals by estimating the room impulse response using an adaptive filter and psychoacoustic frequency scales, reducing memory consumption and improving robustness against noise and fader/balance changes.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If noise dependent equalization systems are used to correct sound distortions in a room, then sound quality is improved, but the system becomes sensitive to background noise which deteriorates signal-to-noise ratio
Solution Approach 1:
The patent converts the harmful background noise into a useful signal by using it as a reference input for the adaptive filter. The noise extraction block processes the microphone signal to separate the room impulse response from the noise, and the adaptive filter uses the noise signal to continuously track and compensate for acoustic changes, thereby improving measurement precision despite noise presence.
Solution Approach 2:
The system implements continuous feedback through the adaptive filter that monitors the microphone signal and adjusts the equalization parameters in real-time. The feedback loop compares the estimated room impulse response with the actual acoustic environment and dynamically adapts to maintain accurate sound correction despite varying noise conditions.
2Measurement precision
If room impulse response data is stored for various fader/balance settings to maintain audio quality, then sound accuracy is improved, but memory consumption increases significantly
Solution Approach 1:
The patent replaces static storage of multiple room impulse response datasets with a dynamic adaptive filter that continuously adjusts its parameters based on current acoustic conditions. Instead of storing fixed data for each fader/balance setting, the system dynamically tracks acoustic changes and adapts the equalization in real-time, significantly reducing memory requirements while maintaining sound accuracy.
Solution Approach 2:
The system changes the approach from storing complete room impulse response datasets to storing and adapting only the essential filter parameters. The adaptive filter modifies its coefficients based on detected acoustic changes, allowing the system to maintain high sound reproduction accuracy while storing minimal data compared to storing full impulse response data for multiple settings.
3Measurement precision
If equalization is used to correct sound distortions caused by room reflections, then timbre accuracy is improved, but the system complexity increases
Solution Approach 1:
The patent merges the noise extraction functionality with the adaptive filter processing in a unified system. The noise extraction block and adaptive filter work together as an integrated signal processing chain that simultaneously handles noise separation and room impulse response estimation, reducing system complexity compared to separate independent systems for each function.
Solution Approach 2:
The adaptive filter serves multiple functions simultaneously: it extracts the room impulse response from the microphone signal, tracks acoustic changes in real-time, and provides continuous equalization adjustment. This multi-functionality eliminates the need for separate dedicated systems for each task, thereby reducing overall system complexity while maintaining high timbre control accuracy.
Data Source
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AI summary
A system and method for automatically controlling the timbre of a sound signal in a listening room are also disclosed, which comprise the following: producing sound in the time domain from a re-transformed electrical sound signal in the time domain, in which an electrical sound signal in the time domain being transformed into electrical sound signal in the frequency domain and the electrical sound signal in the frequency domain being re-transformed into the re-transformed electrical sound signal; generating a total sound signal representative of the total sound in the room, wherein the total sound comprises the sound output from the loudspeaker and the ambient noise in the room; processing the total sound signal to extract an estimated ambient noise signal representing the ambient noise in the room; and adjusting the spectral gain of the electrical sound signal in the frequency domain dependent on the estimated ambient noise signal, the electrical sound signal and a room dependent gain signal. The room dependent gain signal being determined from reference room data and estimated room data.