Signal Filtering

Simple SummaryContent extracted from patent full text and abstract with AI.

This patent describes a novel method and system for filtering signals (such as audio) where filter parameters change between frames (update intervals). It proposes a two-stage filter process at the start of each new interval: the first filter gradually fades out the old parameters, while the second filter gradually fades in the new parameters. This avoids sudden jumps (“discontinuities”) in the filtered output when filter settings change between frames. The result is smoother transitions in time-varying filtering, especially useful for audio codecs and post-processing filters such as long-term postfilters in speech and audio coding.

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

• Audio codec implementations for improving speech and audio quality in telecommunications and media
• Post-filtering in voice-over-IP (VoIP) and video conferencing to reduce artifacts from parameter changes
• Digital audio editing and mastering tools that require smooth filter parameter transitions
• Embedded systems in mobile phones, hearing aids, or smart speakers for real-time audio enhancement
• Broadcast equipment for live or pre-recorded media to ensure seamless audio quality over parameter updates
• Music synthesizers or electronic instruments that involve real-time dynamic filtering of sound

BenefitsContent extracted from patent full text and abstract with AI.

• Reduces or eliminates audible artifacts or discontinuities caused by filter parameter changes between frames
• Lower computational complexity compared to prior techniques like cross-fading, enabling real-time implementations on resource-constrained devices
• Improves subjective audio quality by ensuring smoother transitions in time-varying filters
• Easily adaptable to different types of digital filters (IIR, FIR, LPC, etc.) and use in both encoding and decoding workflows
• Provides parameter-based flexibility, enabling selective filtering or bypassing when changes are small or unnecessary, further optimizing efficiency
• Can be used for both speech and music audio, and in various signal processing domains

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Inventors & Applicants

Patent Abstract

There are discussed methods and systems for filtering an information input signal (11, 11a, x), divided into different update intervals, according to parameters varying with the update intervals, to obtain a filtered output signal (y, 15). A system (10) may comprise: - a first filter unit (12) to filter a first filter input signal (11, x) at least at an initial subinterval ( T l ) in a current update interval ( T ) to obtain a first filter output signal (y', 13), according to parameters associated to the preceding update interval, the first filter unit (12) being configured to change the parameters along at least the initial subinterval ( T l ) from a higher-filtering status to a lower-filtering status; and - a second filter unit (14) to filter a second filter input signal (13), at the initial interval ( T l ), according to parameters associated to the current update interval ( T ) to obtain a second filter output signal (15), the second filter unit (14) being configured to change the parameters along at least the initial subinterval ( T l ) from a lower-filtering status to a higher-filtering status. The first filter input signal (11) is based on the information input signal (x), the first filter output signal (13) is an intermediate signal (y'), the second filter input signal is based on the intermediate signal (y'), and the filtered output signal (y) is based on the second filter output signal (15).