Immersive Group Audio with Dynamic Block Size and Redundancy
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Solution Overview
Problem
Existing audio transmission technologies in group communication sessions, particularly in immersive XR environments, face challenges with latency issues due to an emphasis on audio quality, which can be detrimental in video conferencing, livestreaming, and real-time data sharing experiences.
Innovation Solution
Determine audio redundancy based on network congestion and latency metrics, using a weighted average of packet loss rate and expected delay time to balance audio quality and latency, and implement variable audio block sizes to maintain low latency.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If audio redundancy is increased to improve audio quality and handle packet loss, then audio reliability is improved, but latency increases
Solution Approach 1:
The patent implements dynamic adjustment of audio block sizes and redundancy levels based on real-time network conditions. The system monitors packet loss rates and network stability, then adaptively modifies the audio transmission parameters to balance quality and latency requirements, allowing the audio system to transition between different operational states optimally
Solution Approach 2:
The system changes key transmission parameters including audio block size, redundancy level, and jitter buffer duration based on network conditions. By adjusting these parameters dynamically, the system can reduce latency when network conditions permit while maintaining audio quality through increased redundancy when packet loss is detected
2Manufacturing precision
If audio block size is increased to improve audio quality, then manufacturing precision is improved, but loss of time increases
Solution Approach 1:
The patent makes the audio block size a dynamic parameter that adjusts based on network conditions and latency requirements. Rather than using fixed block sizes, the system can switch between smaller blocks for low-latency applications and larger blocks for high-quality audio transmission when latency is less critical
Solution Approach 2:
The audio stream is segmented into variable-sized blocks that can be optimized for different transmission scenarios. The system can divide audio data into smaller segments when low latency is required and use larger segments when audio quality is the priority, allowing flexible adaptation to different communication needs
3Reliability
If jitter buffer size is increased to handle packet loss, then reliability is improved, but loss of time increases
Solution Approach 1:
The jitter buffer size is implemented as a dynamic parameter that adjusts based on observed network conditions and packet loss patterns. The system monitors the stability of audio packet arrival and adapts the buffer duration accordingly, reducing buffer size to minimize latency when network conditions are stable and increasing it to ensure audio continuity when packet loss occurs
Data Source
AI summary
Aspects of the subject technology may provide low latency audio for group communication sessions. Low latency audio may be provided, some examples, by an electronic device using a lowest audio block size that is lower than a lowest audio block size that is available to one or more other electronic devices in an group communication session.