Audio/Video Stream Sampling for Multi-Stream Jitter Analysis
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Solution Overview
Problem
Existing methods for analyzing real-time audio and video data streams in network environments, such as cross-network video conferencing and IPTV, face inefficiencies in processing load and resource utilization, particularly in handling large numbers of streams and packets, without ensuring real-time service quality.
Innovation Solution
A method and device for analyzing data streams that allocate sampling periods with varying start times to audio and video streams, copying first and last arrival packets within these periods, and calculating sender and receiver delays, jitter, and packet loss rates based on reduced sample data packets, thereby optimizing processing load and identifying network faults.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If all data packets are processed for quality analysis, then measurement precision is improved, but processing load and device complexity increase significantly
Solution Approach 1:
The patent segments the data stream into discrete sampling periods, where only specific packets (first and last arrival packets) within each period are selected for analysis. This segmentation approach reduces the processing load by analyzing only a subset of packets while maintaining quality assessment accuracy through systematic sampling across multiple periods.
Solution Approach 2:
The patent extracts only the essential packets (first and last arrival packets) from each sampling period for quality analysis, rather than processing all packets. This extraction method maintains the key quality metrics (jitter, packet loss, delay) while significantly reducing the processing burden on the analysis device.
2Productivity
If sampling periods are implemented to reduce processing load, then productivity is improved, but measurement precision may deteriorate
Solution Approach 1:
The patent implements preliminary sampling action by pre-defining sampling periods and selecting specific packets (first and last arrival packets) within each period before full analysis is required. This preliminary sampling enables efficient processing while maintaining measurement precision through systematic selection of representative packets that capture key quality characteristics.
Solution Approach 2:
The patent creates a simplified representation (copy) of the data stream quality through sampling packets, which captures the essential quality metrics without requiring full packet processing. This copying approach maintains measurement precision by preserving key quality characteristics while significantly improving processing efficiency.
3Adaptability or versatility
If multiple audio and video streams are analyzed simultaneously, then adaptability is improved, but processing load increases
Solution Approach 1:
The patent segments the analysis of multiple audio and video streams into discrete sampling periods, where each stream is processed independently through systematic sampling. This segmentation enables simultaneous multi-stream analysis while controlling CPU load by processing only the first and last arrival packets from each stream within each sampling period.
Data Source
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AI summary
Provided are a method and a device for analyzing a data stream. The method may comprise: allocating one of a plurality of sampling periods to an audio and video stream, such that different audio and video streams are sampled respectively within the plurality of sampling periods, the plurality of sampling periods having a same sampling duration and having different start times respectively; copying a first arrival data packet of the audio and video stream within the allocated sampling period as a first sample data packet based on a sample stream entry corresponding to the audio and video stream; copying a last arrival data packet of the audio and video stream within the allocated sampling period as a second sample data packet based on the sample stream entry; calculating a sender delay and a receiver delay of the audio and video stream based on respective timestamps of the first sample data packet and the second sample data packet; and analyzing a jitter of the audio and video stream based on the sender delay and the receiver delay.