Multi-Channel Audio Filter Morphing With Stable 3-Axis Interpolation
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Solution Overview
Problem
Conventional digital audio filters struggle with real-time morphing in multiple axes and fail to provide an unconditionally stable structure, as well as independent control of pole and zero frequency and amplitude, leading to undesirable interactions and limitations in dynamic filtering.
Innovation Solution
A multi-channel morphing digital audio filter module that interpolates between multiple fixed frequency responses in real-time using control input signals, decouples frequency and resonance, and generates filter coefficients for a 14th-order filter, ensuring stability and perceptually even changes.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If conventional digital filtering methods are used, then the filter structure is simple, but real-time morphing in multiple axes cannot be achieved and stability cannot be guaranteed
Solution Approach 1:
The filter is divided into multiple independent filter sections, each with its own set of filter coefficients. This segmentation allows each section to be independently controlled and morphed, enabling multi-axis morphing while maintaining overall system stability. The plurality of filter sections can be operated independently or in combination to achieve complex filtering operations.
Solution Approach 2:
The filter employs dynamically adjustable filter coefficients that can be modified in real-time through linear interpolation between predetermined sets of coefficients. This dynamic coefficient adjustment enables smooth transitions between different frequency responses and allows the filter to adapt its characteristics continuously without losing stability.
2Ease of operation
If pole and zero frequency and amplitude are not completely independent, then the filter structure is simpler, but undesirable interactions occur between parameters
Solution Approach 1:
The invention extracts and separates the control of pole frequency, zero frequency, and amplitude parameters into independent control mechanisms. Each parameter can be adjusted independently through separate coefficient sets, eliminating undesirable interactions between parameters while allowing complete independent control of the filter's frequency and amplitude characteristics.
3Productivity
If fixed frequency responses are used, then the filter is stable, but dynamic filtering capabilities are limited
Solution Approach 1:
Multiple predetermined sets of filter coefficients are pre-calculated and stored, each corresponding to a specific frequency response characteristic. These pre-prepared coefficient sets ensure stability while enabling rapid transitions between different filtering modes through linear interpolation, providing both reliability and dynamic capability.
Solution Approach 2:
The filter systematically changes its parameters by interpolating between predetermined coefficient sets based on control signals. This parameter transformation approach allows the filter to dynamically adjust its frequency response while maintaining stability, as the interpolation ensures smooth transitions between known stable states.
4Ease of operation
If smooth interpolation between frequency responses is achieved, then perceptually even changes are provided, but the computational complexity increases
Solution Approach 1:
The invention replaces complex real-time filter design calculations with pre-computed coefficient sets and simple linear interpolation operations. This substitution of heavy computational mechanics with lighter mathematical operations achieves smooth perceptual transitions while significantly reducing the computational power and processing resources required.
Data Source
AI summary
The present disclosure provides a multi-channel morphing digital audio filter, including methods and systems for real-time interpolation between a plurality of fixed frequency responses. Real-time interpolation may be performed simultaneously along three separate axes. Various embodiments of the present technology include receiving, in real-time via a digital filter module, at least one acoustic signal and one or more control input signals indicative of one or more corresponding interpolator values. The methods and systems may further include determining one or more filter coefficients based on a linear interpolation in an encoded space of a plurality of fixed frequency responses, generating a filter from the one or more determined filter coefficients, and applying the generated filter to the at least one received acoustic signal. In some embodiments, frequency and amplitude (i.e., angle and radius) of poles and zeros of the filters may be processed independently, and poles may be unconditionally stable.