Send Queue Audio Streaming Synchronization
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Solution Overview
Problem
Existing streaming technologies face challenges in maintaining audio/video quality due to bandwidth and latency constraints in network connections, leading to congestion and potential network failures, especially in home automation networks where synchronized media playback across multiple rooms is desired.
Innovation Solution
A system and method for audio streaming that utilizes an audio server and client architecture with buffering mechanisms, synchronized clocking through Network Time Protocol (NTP), and the Audio Stream Protocol (ASP) to ensure synchronized audio output across multiple clients, using TCP for network transport to improve reliability and reduce latency.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If streaming content is delivered over network with limited bandwidth and latency, then content delivery is achieved, but audio/video quality and synchronization are degraded
Solution Approach 1:
The patent implements a send queue that pre-buffers audio data packets before transmission. This preliminary action ensures that even when network bandwidth is limited or latency occurs, the client device has packets ready to play, maintaining both streaming reliability and audio quality without interruption or desynchronization.
Solution Approach 2:
The patent uses buffering mechanisms at both the server (send queue) and client sides to create a time cushion against network failures and latency. This beforehand cushioning protects the streaming system from network issues, ensuring continuous playback with maintained quality and synchronization across multiple clients.
2Speed
If streaming content is delivered without buffering, then latency is reduced, but network failures and quality degradation occur
Solution Approach 1:
The send queue pre-prepares and queues audio packets for transmission, creating a buffer that allows immediate playback when packets arrive at the client. This preliminary action reduces the need for large client-side buffers while maintaining reliability, achieving fast startup and continuous playback even with network variability.
3Productivity
If multiple clients receive streams simultaneously without synchronization, then content delivery is achieved, but noticeable delays occur across rooms
Solution Approach 1:
The patent implements synchronized clocking using Network Time Protocol (NTP) where all clients continuously synchronize their clocks with the server and each other. This feedback mechanism ensures that playback timestamps are coordinated across all clients, maintaining synchronization accuracy within a few milliseconds even as multiple clients receive and play back content simultaneously across different locations.
4Reliability
If TCP protocol is used for network transport, then reliability is improved, but latency may increase
Solution Approach 1:
The send queue pre-buffers multiple audio packets before transmission over TCP. This preliminary action allows the system to absorb TCP's inherent latency and retransmission delays, ensuring that playback is not interrupted by network delays. The buffer acts as a cushion that maintains both TCP's reliability benefits while mitigating its latency impact.
Data Source
AI summary
A system and method are provided for preparing a streaming media system for initial presentation of a media stream. The system includes a media server configured to send out media packets for a media stream at periodic clocked intervals. A framer can be located with the media server to divide the media stream into media packets. A media client is also provided to receive the media packets for the media stream from the media server. A send queue can be located in the server. The send queue can be configured to store a defined length of programming from the media stream, and the send queue can immediately fill a client's user buffer when an activation event occurs.