Audio Processing for Speech Using Late Reverberation Subtraction
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
End devices face challenges in accurately processing voice input due to reverberation, which increases error rates and degrades speech signal quality, especially in environments where reflections of the user's voice interfere with the direct signal.
Innovation Solution
An audio processing service that estimates and subtracts late reverberation using adaptive filtering techniques, either linearly or non-linearly, to improve speech recognition and speech-to-text accuracy by distinguishing between direct and reflected voice signals.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If reverberation cancellation is not applied, then the system operates with simple processing, but speech recognition accuracy deteriorates due to reflected voice signals
Solution Approach 1:
The system performs preliminary action by estimating the room impulse response and calculating late reverberation components before speech recognition occurs. This allows the reverberation to be removed in advance from the speech signal, improving recognition accuracy without adding complexity during the critical recognition phase
Solution Approach 2:
The speech signal is segmented into direct speech components and reverberation components based on temporal characteristics. By dividing the signal processing into distinct segments (direct path vs. reflected paths), the system can selectively remove late reverberation while preserving the direct speech signal
2Reliability
If late reverberation is removed using adaptive filtering, then speech signal quality improves, but computational requirements increase
Solution Approach 1:
The system applies partial action by selectively removing only the late reverberation portion of the signal rather than processing the entire signal equally. By targeting only the harmful late reverberation components and leaving early reverberation and direct speech untouched, computational energy is conserved while still improving signal quality
Solution Approach 2:
The adaptive filter uses the speech signal itself to generate the room impulse response estimate through correlation operations. This self-service approach eliminates the need for external measurement equipment or additional sensors, reducing computational overhead while maintaining reliability
3Measurement precision
If room impulse response estimation is performed continuously, then reverberation cancellation accuracy improves, but processing time increases
Solution Approach 1:
The system performs room impulse response estimation periodically rather than continuously, updating the estimate at intervals during speech activity. This periodic approach maintains adequate cancellation accuracy while significantly reducing processing time and computational load compared to continuous estimation
Solution Approach 2:
The system skips redundant estimation operations by detecting speech activity and only performing impulse response estimation when speech is detected. During non-speech periods, the system rushes through with minimal processing, maintaining accuracy during critical speech segments while minimizing time loss during inactive periods
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 minimizes error rates and enhances the quality of speech input signals by effectively canceling out late reverberation, leading to improved speech recognition and reduced word/sentence error rates.
Implementation Method 1
the end device may receive voice input not only from the user but also from reflections (e.g., reverberation) of the user's voice that stem from the environment
Implementation Method 2
An audio processing service that estimates and subtracts late reverberation using adaptive filtering techniques
Data Source
AI summary
A method, a device, and a non-transitory storage medium are described in which a power of late reverberation of a speech signal is estimated based on early samples of the speech signal. The power of the late reverberation may be subtracted linearly or non-linearly from the speech signal.