Dynamic Audio Video Buffering for Network Jitter
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Solution Overview
Problem
Real-time audio and video communication systems face challenges in balancing latency and quality due to resource constraints, particularly in packet-switched networks like the Internet, where increased quality leads to higher processing demands and potential jitter, causing delays and distortion in data transmission.
Innovation Solution
A system where a network terminal dynamically adjusts audio and video encoding and buffering parameters based on system performance, network conditions, and user preferences to maintain optimal quality and minimize latency, using modules like system monitor, control, audio translator, video translator, and network translator to manage data transmission effectively.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Manufacturing precision
If audio and video quality is increased, then communication quality is improved, but system resources are consumed more and latency increases
Solution Approach 1:
The system dynamically adjusts audio and video encoding parameters based on real-time monitoring of system performance and network conditions. The control module modifies buffering parameters, encoding rates, and quality settings during operation to balance quality and latency requirements
Solution Approach 2:
The system changes encoding parameters such as bit rate, sampling rate, and buffer size based on detected system conditions. When resources are constrained or latency increases, the system reduces encoding quality parameters to maintain acceptable performance
2Reliability
If buffer size is increased to compensate for jitter, then audio presentation stability is improved, but additional latency is introduced
Solution Approach 1:
The buffering parameter is dynamically adjusted based on real-time network conditions and system performance. The control module monitors jitter and resource consumption to determine optimal buffer sizes that stabilize audio presentation without introducing excessive latency
Solution Approach 2:
The system uses feedback from system performance monitoring and network condition detection to continuously optimize buffering parameters. The control module receives data about actual performance and adjusts buffer settings to achieve stability without unnecessary delay
3Manufacturing precision
If processing capacity is increased to handle higher quality data, then audio/video quality is improved, but system resource consumption increases
Solution Approach 1:
The system changes encoding and processing parameters based on available system resources and detected performance constraints. When processing capacity is limited, the system reduces quality parameters to maintain acceptable audio/video output without exhausting resources
Data Source
AI summary
Systems and methods are described for transmitting audio and video messages between network terminals over a network. A communication link is established between a first network terminal and a second network terminal, and audio data and video data is transmitted concurrently over the link. During communication, the first network terminal acquires data representing the performance of the first network terminal and data representing the performance of the network. Based on negotiation between the first network terminal and the second network terminal and received performance data the at least one buffering parameter is determined and, if the parameter is within a predetermined range, the buffering parameter of the first network terminal is modified.