All-Pass Audio Peak Reduction Without Harmonic Distortion

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Solution Overview

Problem

Traditional audio signal processing methods for dynamic range reduction introduce nonlinear techniques that result in harmonic distortion, affecting sound quality.

Innovation Solution

A method using an all-pass filter with a synced delay parameter and gain parameter, optimized through gradient descent and linear programming, to reduce audio peaks while minimizing computational complexity and distortion.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Reliability

If traditional nonlinear techniques are used for dynamic range reduction, then peak amplitude limiting is achieved, but harmonic distortion is introduced that negatively affects sound quality

Engineering Contradiction:
Improvepeak amplitude limiting effectivenessVSAvoidharmonic distortion
Core Design Contradiction:
ReliabilityVSObject-generated harmful factors

Solution Approach 1:

The patent changes the fundamental parameter of the compression approach from nonlinear to linear. By using a linear all-pass filter with carefully designed impulse response instead of traditional nonlinear clipping or limiting, the system achieves peak reduction without introducing harmonic distortion. The linear operation preserves the spectral content while redistributing energy in time.

Inventive Principle:
Principle #35Parameter changes

Solution Approach 2:

The patent replaces the mechanical nonlinear limiting mechanism with an electronic linear filtering approach. Instead of using hard clipping or soft knee limiting that inherently create distortion, the invention uses a linear convolution operation with a pre-computed impulse response that achieves the same peak control goal without spectral contamination.

Inventive Principle:
Principle #28Mechanics substitution (Replace mechanical system)

2Object-generated harmful factors

If computationally intensive methods are used for peak reduction, then distortion is minimized, but calculation complexity increases

Engineering Contradiction:
Improvedistortion levelVSAvoidcomputational complexity
Core Design Contradiction:
Object-generated harmful factorsVSDevice complexity

Solution Approach 1:

The patent performs preliminary computation of the all-pass filter impulse response offline or in advance, storing it for repeated use. This pre-computation allows the real-time processing to simply convolve the input signal with the pre-designed impulse response, significantly reducing online computational complexity while maintaining the distortion-free linear processing advantage.

Inventive Principle:
Principle #10Preliminary action

Solution Approach 2:

The patent segments the peak reduction problem into two independent parts: (1) designing the all-pass filter impulse response characteristics offline, and (2) applying the filter in real-time through simple convolution. This segmentation allows complex optimization to be done once, while real-time processing remains computationally lightweight.

Inventive Principle:
Principle #1Segmentation

Data Source

PatentUS11881829B2Method for audio peak reduction using all-pass filter
Publication Date: 2024.01.23 AAC MICROTECH (CHANGZHOU) CO LTD
  • US11881829B2 patent drawing
  • US11881829B2 patent drawing

AI summary

Provided is a method for audio peak reduction using an all-pass filter, including: determining a delay parameter m and a gain parameter g based on a formula (1):arg⁢ minm,gmaxn<semantics definitionURL="">❘<annotation encoding="Mathematica">"\[LeftBracketingBar]"</annotation></semantics>ym,g(n)<semantics definitionURL="">❘<annotation encoding="Mathematica">"\[RightBracketingBar]"</annotation></semantics>,(1)absolute peak mapY⁡(m,g)=maxn<semantics definitionURL="">❘<annotation encoding="Mathematica">"\[LeftBracketingBar]"</annotation></semantics>ym,g(n)<semantics definitionURL="">❘<annotation encoding="Mathematica">"\[RightBracketingBar]"</annotation></semantics>,ym,g(n) represents a processed signal with a time-domain response function and is calculated based on a formula (2): ym,g(n)=(hs*x)(n)(2), where hs represents an impulse response function, x (n) represents an input signal, and hs is calculated based on formula (3):HS(z)=g+z-m1+gz-m.(3)This method is widely used in the reproduction, storage and broadcasting of sound, and the computational complexity is small, which is a supplement to the traditional nonlinear compression algorithm.