Smart Audio Coding Mode Switching for Packet Loss Mitigation
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Existing voice communication technologies face challenges in maintaining high-quality audio transmission due to packet loss and network conditions, leading to increased end-to-end delay and reduced user experience, as conventional FEC schemes either introduce additional delay or compromise baseline speech quality.
Innovation Solution
A device and method that dynamically adjust the coding mode by switching between primary frame encoding with and without redundant frame encoding, based on network conditions and real-time quality metrics, using a de-jitter buffer and analyzer to determine when to request a coding mode change from a channel-aware mode to a non-channel-aware mode, thereby optimizing speech quality, delay, and data rate.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If the buffer depth of the de-jitter buffer is increased, then the quality of the processed audio signal improves, but the end-to-end delay increases
Solution Approach 1:
The patent implements dynamic buffer depth adjustment where the de-jitter buffer adapts its depth based on real-time network conditions. When network conditions are good, the buffer depth is reduced to minimize delay. When packet loss increases, the buffer depth is increased to allow more packets to be collected for error recovery, thus dynamically optimizing the trade-off between quality and delay.
Solution Approach 2:
The system changes the buffer depth parameter adaptively based on channel conditions. The analyzer monitors network conditions and triggers buffer depth adjustments when specific conditions are met, allowing the system to optimize audio quality by collecting more packets when needed while maintaining low latency when possible.
2Loss of time
If the buffer depth of the de-jitter buffer is reduced, then the end-to-end delay decreases, but the quality of the processed audio signal deteriorates
Solution Approach 1:
The buffer operates dynamically with adjustable depth rather than a fixed value. This allows the system to reduce buffer depth to minimize delay when network conditions permit, while having the capability to increase depth when quality requirements demand it, based on real-time analyzer feedback.
Solution Approach 2:
The buffer depth parameter is changed adaptively based on channel conditions. When the analyzer determines that network conditions are favorable, the buffer depth is reduced to achieve lower latency. When conditions deteriorate, the parameter is adjusted upward to maintain audio quality.
3Reliability
If redundant frame encoding is used, then speech quality under packet loss improves, but data rate increases
Solution Approach 1:
Instead of always transmitting full redundant frames, the system uses partial frame encoding where only essential portions of frames are transmitted as redundancy. This provides error recovery capability while using fewer bits than complete redundant frame transmission, thus improving speech quality under packet loss with a more moderate increase in data rate.
Solution Approach 2:
The coding mode parameter is changed dynamically between different encoding schemes. The system can switch between modes with and without redundant frame encoding based on network conditions, allowing optimization of the trade-off between speech quality reliability and data rate consumption.
4Reliability
If coding mode is changed dynamically, then optimization of speech quality and delay is achieved, but system complexity increases
Solution Approach 1:
The system implements a feedback mechanism where the analyzer monitors network conditions and provides information to trigger coding mode changes. This feedback loop allows the system to automatically adapt to changing conditions and optimize speech quality without requiring complex manual control or intervention.
Solution Approach 2:
The de-jitter buffer and analyzer work together in a self-service manner where the analyzer automatically detects when coding mode changes are needed based on predefined conditions, and the buffer self-adjusts its operation mode without external intervention, reducing the complexity of external control systems.
Data Source
AI summary
A method of smart coding mode switching includes receiving a first data including a primary copy and a partial copy. The method includes determining if switching a coding mode from channel aware mode to non-channel aware mode may be advantageous. The method further includes transmitting a request to another device for coding mode switch in response to determination result. The method includes receiving and decoding of a second data that includes a primary copy.