Spatial Audio Metadata Synchronization via Jitter Buffer Adaptation
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Immersive audio codecs face challenges in maintaining synchrony between spatial audio metadata and transport audio signals, leading to unwanted artefacts and decreased audio quality due to network jitter and packet loss, especially in packet-based networks like 4G and 5G networks.
Innovation Solution
The proposed solution involves time-adapting spatial audio metadata to maintain synchrony with the transport audio signals by using a metadata adaptor that adjusts spatial audio metadata parameters based on time adjustment instructions from the adaptation control logic processor, ensuring that the metadata remains in sync with the time-adjusted transport audio signals.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If spatial audio metadata is transmitted through packet-based networks (4G/5G), then network coverage and accessibility are improved, but synchronization between metadata and transport audio signals deteriorates due to network jitter and packet loss
Solution Approach 1:
The patent applies preliminary action by predicting future synchronization states based on historical jitter buffer delay data. The system estimates upcoming synchronization issues before they occur and proactively adjusts metadata timing parameters, rather than reacting after desynchronization has already degraded audio quality. This predictive approach allows the system to compensate for network jitter effects in advance.
Solution Approach 2:
The patent implements feedback mechanisms by continuously monitoring actual synchronization performance between spatial audio metadata and transport audio signals. The system uses this feedback information to dynamically adjust jitter buffer delay parameters and metadata timing, creating a closed-loop control system that adapts to changing network conditions and maintains synchronization reliability.
2Reliability
If jitter buffer delay is increased to compensate for network jitter, then synchronization reliability is improved, but audio latency increases
Solution Approach 1:
The patent applies dynamics by making the jitter buffer delay parameter adaptive rather than static. The system dynamically adjusts the delay based on real-time network conditions, audio content characteristics, and synchronization performance metrics. This allows the system to increase delay only when necessary for synchronization while minimizing latency under normal conditions, achieving an optimal balance between reliability and time efficiency.
Solution Approach 2:
The patent changes key parameters including jitter buffer delay duration, metadata timing offsets, and resynchronization thresholds based on operating conditions. By dynamically modifying these parameters rather than using fixed values, the system can optimize the trade-off between synchronization reliability and audio latency for different network scenarios and content types.
3Manufacturing precision
If spatial audio metadata is resynchronized with transport audio signals, then audio quality is maintained, but processing complexity increases
Solution Approach 1:
The patent applies local quality by selectively resynchronizing only the spatial audio metadata that requires it, rather than processing the entire audio stream uniformly. The system identifies specific metadata parameters and time segments that need synchronization adjustments and applies processing only to those localized portions, preserving audio quality while minimizing overall processing complexity.
Solution Approach 2:
The patent extracts and separates the synchronization adjustment process from the main audio decoding pipeline. By isolating the metadata resynchronization as a distinct, modular operation, the system can apply sophisticated synchronization algorithms without burdening the entire audio processing chain, thereby maintaining audio quality while managing processing complexity through functional decomposition.
Data Source
Figure 1
Figure 2
Figure 3
AI summary
An apparatus for spatial audio decoding, configured to: receive a first audio signal frame comprising a number of subframes; receive a parameter indicating a total number of time slots of a second audio signal; mapping the total number of time slots of the second audio signal frame to the first number of subframes of the first audio signal frame to produce a map for mapping a time slot of the second audio signal to a subframe of the first audio signal; using this map for mapping a time slot of the second audio signal to a subframe of the first audio signal to produce a map for mapping a subframe of the second audio signal to a subframe of the first audio signal; and using the map for mapping a subframe of the second audio signal to a subframe of the first audio signal to assign at least one spatial audio parameter of a subframe of the first audio signal to a subframe of the second audio signal.