Time Domain Aliasing Reduction for Non-Uniform Filterbanks Which Use Spectral Analysis Followed by Partial Synthesis

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This patent describes an improved technique for processing audio signals using non-uniform filterbanks based on modified discrete cosine transforms (MDCTs). Specifically, it introduces a method and processor for reducing aliasing—an unwanted artifact—in the time domain when creating subband representations of audio. The approach uses a cascaded lapped transform (e.g., multi-stage MDCT) on overlapping audio blocks, followed by a time domain aliasing reduction stage that combines overlapping subband samples in a weighted manner to minimize aliasing effects. This results in more accurate and temporally compact representations of audio subbands with low delay and no added redundancy.

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

• Audio encoding and decoding in digital music streaming services for higher sound quality.
• Audio signal processing in hearing aids or cochlear implants to provide clearer sound perception.
• Speech codecs for VoIP, telephony, or conferencing systems to deliver better voice intelligibility.
• Acoustic scene analysis, such as automatic music transcription or sound event detection, using subband features.
• Noise reduction and audio enhancement algorithms for broadcasting or professional audio production.
• Embedded audio analysis modules for smart devices (e.g., smart speakers, phones).

BenefitsContent extracted from patent full text and abstract with AI.

• Reduces time-domain aliasing, improving audio quality by minimizing artifacts.
• Enables the creation of non-uniform, perceptually motivated subbands for more flexible signal representation.
• Maintains low computational delay, making it suitable for real-time audio applications.
• Allows arbitrary frequency scales and window configurations, providing high adaptability.
• No additional data redundancy is introduced, keeping data size and processing efficient.
• The approach integrates smoothly with standard MDCT-based coding pipelines, easing adoption in existing systems.

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Patent Abstract

Embodiments provide an audio processor for processing an audio signal to obtain a subband representation of the audio signal. The audio processor comprises a cascaded lapped critically sampled transform stage and a time domain aliasing reduction stage. The cascaded lapped critically sampled transform stage is configured to perform a cascaded lapped critically sampled transform on at least two partially overlapping blocks of samples of the audio signal, to obtain a set of subband samples on the basis of a first block of samples of the audio signal, and to obtain a corresponding set of subband samples on the basis of a second block of samples of the audio signal. The time domain aliasing reduction stage is configured to perform a weighted combination of two corresponding sets of subband samples, one obtained on the basis of the first block of samples of the audio signal and one obtained on the basis on the second block of samples of the audio signal, to obtain an aliasing reduced subband representation of the audio signal.