Audio Processor, Audio Processing System, Audio Decoder, Method for Providing a Processed Audio Signal Representation and Computer Program Using a Time Scale Modification

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This invention relates to an audio processor and method for improving audio signal handling, especially in communication systems and immersive audio applications. The core idea is to apply a time scale modification (adjusting speed or duration of the audio signal) at an intermediate stage during audio processing, rather than only before or after the main processing steps. By modifying the timing of intermediate audio signals (like decoded transport channels) before rendering or upmixing, the system can achieve smoother playback with lower latency, better adaptability to network jitter, and improved audio quality, particularly when reconstructing complex multi-channel or scene-based audio.

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

• Voice-over-IP (VoIP) or telecommunication systems to ensure consistent audio playout despite network delays and jitter.
• Immersive audio applications (such as AR/VR) where real-time adjustment to listener movement and rapid rendering is required.
• Multi-channel audio streaming or conferencing systems.
• Speech recognition systems that require temporally consistent audio input.
• Broadcast or streaming platforms that need to adapt audio in real-time for synchronization or user interaction.
• Professional audio workstations or DAWs for non-destructive time-stretching during editing or mixing.

BenefitsContent extracted from patent full text and abstract with AI.

• Significantly reduces audio playback latency and optimizes synchronization between user actions (like head movements) and sound output.
• Minimizes computational complexity by applying time scale modification on fewer intermediate channels before upmixing or rendering, rather than on all output channels.
• Improves audio quality by allowing precise adaptation of processing parameters and metadata to match the altered timing, reducing artifacts or discontinuities.
• Enhances robustness against network jitter or packet delay, reducing interruptions or glitches during streaming or real-time communication.
• Provides flexibility and scalability for complex audio scenes, enabling efficient, real-time rendering in distributed or edge devices.
• Enables quick response to user interaction, as time scaling is managed before rendering or localization adjustments.

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Patent Abstract

An audio processor for providing a processed audio signal representation on the basis of an input audio signal representation performs a plurality of processing steps, in order to provide the processed audio signal representation on the basis of the input audio signal representation. The audio processor performs a time scale modification on one or more intermediate audio signals, which are provided by a first processing, in order to obtain one or more time-scale-modified intermediate audio signals, and the audio processor performs a second processing, which follows the first processing, on the basis of the one or more time-scale-modified intermediate audio signals. An audio processing system, a method and a computer program are also described.