Room Impulse Response Filter Length Adaptation
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Audio systems face challenges in accurately modeling the effect of a recording space on audio signals, particularly due to differences between near-field and far-field microphones, leading to audible artefacts and non-natural sounding audio content, as near-field signals are less affected by the recording space while far-field signals are influenced by reverberation and ambient sounds.
Innovation Solution
The apparatus determines the filter length of a room impulse response filter, with a first portion having a direct acoustic propagation delay consistent across frequency bands and a second portion being frequency-dependent, using methods such as direct-path delay estimation and reverberation time analysis, to accurately model the recording space and adjust filter lengths based on signal volume and system geometry.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Device complexity
If a single filter length is used across all frequency bands, then the device complexity is reduced, but the manufacturing precision of audio modeling deteriorates
Solution Approach 1:
The filter is divided into multiple frequency bands (e.g., low, mid, high frequencies) with different filter lengths. Each frequency band has its own optimized filter length parameter, allowing precise modeling of acoustic properties at different frequencies while maintaining manageable system complexity through structured segmentation.
Solution Approach 2:
Different portions of the frequency spectrum are assigned different filter lengths based on their specific acoustic characteristics. Low frequencies use longer filter lengths to capture reverberation, while high frequencies use shorter lengths to avoid noise amplification, optimizing the local quality of audio modeling for each frequency region.
2Manufacturing precision
If frequency-dependent filter lengths are used, then the manufacturing precision of audio modeling is improved, but the device complexity increases
Solution Approach 1:
The filter length is made dynamic and adaptive rather than static. The system automatically adjusts filter lengths based on detected acoustic environment characteristics, signal energy levels, and frequency content, allowing the filter configuration to adapt dynamically to different recording conditions without manual intervention.
Solution Approach 2:
The filter length parameter is changed based on frequency band and acoustic conditions. The system modifies the filter length parameter dynamically according to the specific frequency range and detected environmental properties, optimizing modeling accuracy for each condition while avoiding the need for complex manual configuration.
3Manufacturing precision
If longer filter lengths are used to capture reverberation, then the manufacturing precision of space modeling is improved, but the loss of time increases
Solution Approach 1:
Instead of using uniformly long filters for all frequencies, the system applies partial action by using longer filter lengths only where necessary (low frequencies with significant reverberation) and shorter lengths where reverberation is minimal (high frequencies). This reduces unnecessary computational processing time while maintaining space modeling accuracy where it matters most.
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
This approach improves audio quality by reducing artefacts and computational load, providing more accurate modeling of the recording space and enhancing the subjective performance of audio mixing by adapting filter lengths to the specific acoustic properties and source characteristics.
Implementation Method 1
means (such as a near-field microphone) for receiving a near-field audio source signal from a near-field microphone
Implementation Method 2
means (such as an array of one or more far-field microphones) for receiving a far-field audio signal from an array comprising one or more far-field microphones
Implementation Method 3
The effect of a recording space on array can be modelled using one or more room impulse response filters (RIRs)
Data Source
AI summary
An apparatus, method and computer program is described comprising: receiving a near-field audio source signal from a near-field microphone (22); receiving a far-field audio signal from an array comprising one or more far-field microphones (23); determining a filter length of a first portion of a room impulse response filter for the near-field microphone, wherein said filter length of said first portion is the same at each of a plurality of frequency bands of the filter and wherein said filter length of said first portion includes a direct acoustic propagation delay; and determining a filter length of a second portion of the room impulse response filter at each of the plurality of frequency bands, wherein the filter length of said second portion is frequency-dependent.