Signal Processor and Method for Providing a Processed Audio Signal Reducing Noise and Reverberation

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This invention describes a signal processor and method for improving the quality of audio signals by reducing both noise and reverberation (echo or lingering sounds) in real time. The processor uses an advanced mathematical model (autoregressive reverberation model) and parallel Kalman filters to accurately estimate and subtract unwanted noise and reverberation from single or multi-channel audio, such as recordings with multiple microphones. The system allows flexible adjustment of how much noise or reverberation is removed, and can adapt to changing acoustic environments.

Use CasesContent extracted from patent full text and abstract with AI.

• Enhancing speech clarity in teleconferencing and videoconferencing systems, especially in noisy or echoic rooms.
• Improving intelligibility of voice commands in smart home devices and virtual assistants.
• Supporting hearing aids and assistive listening devices to deliver clearer sound in reverberant or noisy spaces.
• Boosting speech recognition accuracy in public address systems, automated transcription, or voice-controlled systems.
• Audio enhancement in surveillance, broadcasting, or archival of recorded materials from challenging acoustic environments.
• Improving communication quality in vehicles, such as cars, airplanes, or trains, with complex noise and echo patterns.

BenefitsContent extracted from patent full text and abstract with AI.

• Simultaneously reduces both noise and reverberation for much clearer audio.
• Operates in real time, suitable for live scenarios and adaptive to changing conditions.
• Works with single or multiple microphones, providing flexibility for various devices or setups.
• Allows independent control over the reduction levels of noise and reverberation, letting users fine-tune for optimal perceptual quality.
• Enables advanced preprocessing for downstream applications like beamforming, speech recognition, or further audio enhancement.
• Improves audio experience and intelligibility for listeners, helping people understand speech better in difficult environments.

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Patent Abstract

A signal processor for providing one or more processed audio signals on the basis of one or more input audio signals is configured to estimate coefficients of an autoregressive reverberation model using the input audio signals and the delayed noise-reduced reverberant signals obtained using a noise reduction. The signal processor is configured to provide noise-reduced reverberant signals using the input audio signals and the estimated coefficients of the autoregressive reverberation model. The signal processor is configured to derive noise-reduced and reverberation-reduced output signals using the noise-reduced reverberant signals and the estimated coefficients of the autoregressive reverberation model. A method and a computer program comprise a similar functionality.