Single-Channel Distant Speech Processing With Virtual Channels
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Solution Overview
Problem
Speech processing systems using a single microphone suffer from recognition accuracy degradation due to far-field speech signals, which are corrupted by room reverberation and background noise, and conventional methods introduce artifacts or fail to effectively utilize single-channel data.
Innovation Solution
Generate multiple microphone representations from a single microphone channel by segmenting the signal into speaker-specific and acoustic parameter-specific signals, applying masks to attenuate unwanted components, and combine these with the original signal using self-attention models to enhance speech processing.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Device complexity
If a single microphone is used, then the device complexity is reduced, but the speech processing accuracy and robustness against noise deteriorate
Solution Approach 1:
The patent segments the single microphone signal into multiple virtual channels by separating speech and noise components in different time-frequency regions. This creates artificial multi-channel data from single-channel input, enabling multi-channel processing techniques to be applied without physical multiple microphones, thus resolving the contradiction between device simplicity and processing accuracy
Solution Approach 2:
The patent transforms the single-channel signal into a multi-dimensional representation by creating time-frequency masks and generating virtual multi-channel data. This dimensional transformation allows the system to exploit spatial processing techniques normally requiring multiple microphones, thereby improving speech accuracy while maintaining single-microphone simplicity
2Object-affected harmful factors
If conventional speech enhancement methods are applied to single channel signals, then some noise reduction is achieved, but artifacts are introduced and speech quality deteriorates
Solution Approach 1:
The patent applies different processing strategies to different time-frequency regions of the signal. By creating local speech and noise masks that adapt to local signal characteristics, the system enhances speech in speech-dominated regions while suppressing noise in noise-dominated regions, avoiding uniform processing that introduces artifacts and maintains speech quality
Solution Approach 2:
The patent introduces time-frequency masks as intermediary structures that mediate between the raw single-channel signal and the enhanced output. These masks act as selective filters that preserve speech components while attenuating noise components, avoiding direct signal manipulation that would introduce artifacts and maintaining speech naturalness
3Reliability
If multi-channel speech processing systems are used, then robustness against noise and reverberation is improved, but the device complexity and cost increase
Solution Approach 1:
The patent creates virtual copies of multi-channel data from a single physical microphone channel. By generating synthetic multi-channel representations through time-frequency analysis and mask-based separation, the system replicates the functionality of physical microphone arrays without the associated hardware complexity and cost, achieving noise robustness through computational rather than physical means
Data Source
AI summary
A method, computer program product, and computing system for receiving a signal from a single microphone. A plurality of modified signals may be generated from the single microphone signal, where the plurality of modified signals include at least one of: a speaker-specific signal, an acoustic parameter-specific signal, and a speech enhanced signal. Speech processing may be performed on the plurality of modified signals.


