Speech Enhancement Model Using Multi-Source Auxiliary Information

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Solution Overview

Problem

Current speech enhancement algorithms face limitations in real-time performance due to fixed window lengths in short-time Fourier transforms and fail to fully utilize rich prior information from multiple sound sources, leading to suboptimal speech quality and intelligibility.

Innovation Solution

A general speech enhancement method using multi-source auxiliary information, comprising a speech enhancement model with an encoder module, attention module, and decoder module, which leverages a database of sound source information to process noisy signals alongside target and interference sound sources, enhancing speech quality through attention modeling and fusion masks.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Measurement precision

If short-time Fourier transform with fixed window length is used for extracting spectrum features, then spectrum features can be extracted, but real-time performance is degraded

Engineering Contradiction:
Improvespectrum feature extractionVSAvoidreal-time performance
Core Design Contradiction:
Measurement precisionVSProductivity

Solution Approach 1:

The patent replaces the traditional short-time Fourier transform (STFT) mechanical processing method with a deep learning-based spectrum feature extraction approach. The neural network model directly processes noisy speech signals to extract spectrum features without requiring fixed window length transformations, thereby eliminating the real-time performance degradation while maintaining accurate spectrum feature extraction capability.

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

Solution Approach 2:

The patent changes the fundamental parameter of signal processing by moving from fixed window length STFT to a flexible deep learning approach. The model adapts to varying signal characteristics dynamically, allowing spectrum feature extraction without being constrained by fixed temporal windows, thus improving real-time performance while preserving measurement precision.

Inventive Principle:
Principle #35Parameter changes

2Measurement precision

If manual spectrum feature design is used, then spectrum features can be extracted for speech enhancement, but the features may not be perfectly suited for speech enhancement tasks

Engineering Contradiction:
Improvespectrum feature extractionVSAvoidspeech enhancement effectiveness
Core Design Contradiction:
Measurement precisionVSReliability

Solution Approach 1:

The patent implements self-service by enabling the neural network model to automatically learn and extract optimal spectrum features directly from noisy speech signals for the specific task of speech enhancement. Instead of relying on manual feature design, the model adapts to the specific characteristics of the enhancement task, ensuring that the extracted features are perfectly suited for improving speech quality and intelligibility.

Inventive Principle:
Principle #25Self-service

Solution Approach 2:

The patent incorporates feedback mechanisms through supervised learning where the model is trained with noisy speech as input and clean speech as target. The loss function provides continuous feedback during training, allowing the model to iteratively improve its spectrum feature extraction capability specifically for speech enhancement tasks, thereby ensuring high reliability and effectiveness.

Inventive Principle:
Principle #23Feedback

3Device complexity

If historical information of sound sources is not utilized, then the algorithm is simple, but speech enhancement performance is suboptimal

Engineering Contradiction:
Improvealgorithm simplicityVSAvoidspeech enhancement performance
Core Design Contradiction:
Device complexityVSReliability

Solution Approach 1:

The patent applies preliminary action by collecting and storing historical sound source information (including target speaker audio, noise audio, and reverberation data) in advance in a sound source information database. This pre-collected information is then utilized during the speech enhancement process to improve performance without significantly increasing algorithmic complexity, as the heavy lifting of data collection is performed beforehand.

Inventive Principle:
Principle #10Preliminary action

Solution Approach 2:

The patent introduces a sound source information database as an intermediary between the noisy speech input and the enhancement process. This database stores pre-collected historical information about sound sources, which acts as a mediator to provide contextual information that improves speech enhancement performance without requiring complex real-time analysis of sound source characteristics.

Inventive Principle:
Principle #24Intermediary (Mediator)

Data Source

PatentUS12094484B2General speech enhancement method and apparatus using multi-source auxiliary information
Publication Date: 2024.09.17 ZHEJIANG LAB
  • US12094484B2 patent drawing
  • US12094484B2 patent drawing
  • US12094484B2 patent drawing

AI summary

The present disclosure discloses a general speech enhancement method and apparatus using multi-source auxiliary information. The method includes following steps: S1: building a training data set; S2: using the training data set to learn network parameters of a model, and building a speech enhancement model; S3: building a sound source information database in a pre-collection or on-site collection mode; S4: acquiring an input of the speech enhancement model; and S5: taking a noisy original signal as a main input of the speech enhancement model, taking auxiliary sound signals of a target source group and auxiliary sound signals of an interference source group as side inputs of the speech enhancement model for speech enhancement, and obtaining an enhanced speech signal.