Low Latency Audio Delivery via Priority Tagged Switched Ethernet
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Solution Overview
Problem
Existing digital audio transmission over packet networks faces challenges with high latency and interference from data signals, leading to inefficient use of bandwidth and poor audio fidelity, especially in environments requiring low latency like broadcast studios.
Innovation Solution
The use of switched Ethernet with priority tagging for audio packets, synchronized clock systems, and optimized buffering to ensure low latency and reliable audio transmission, allowing audio and data to share network links without interference.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Productivity
If audio signals are transmitted over packet networks using standard Ethernet, then network infrastructure cost is reduced and bandwidth is increased, but latency increases and audio fidelity deteriorates due to packet reassembly delays and interference with data traffic
Solution Approach 1:
The patent segments audio traffic into separate virtual channels (VC 1, VC 2, etc.) within the packet network, allowing audio packets to be isolated from data traffic. This segmentation enables audio to receive dedicated bandwidth while maintaining the packet network's overall capacity, resolving the contradiction between high bandwidth utilization and low latency.
Solution Approach 2:
The patent implements preliminary clock synchronization before audio transmission begins. By establishing synchronized timing references in advance and using timestamps in packet headers, the system pre-calculates optimal transmission timing, eliminating the need for complex real-time adjustments and reducing latency during actual audio playback.
2Device complexity
If audio packets share network links with data packets, then network infrastructure is simplified and cost reduced, but audio quality deteriorates due to interference and variable priority treatment
Solution Approach 1:
The patent applies local quality differentiation by assigning different priority levels to different virtual channels. Audio packets in VC 1 receive highest priority, while data packets receive lower priority. This local quality distinction ensures that audio transmission remains reliable and high-fidelity even while sharing the same physical network infrastructure with data traffic.
Solution Approach 2:
The patent introduces virtual channels as intermediary structures between the physical network and the audio/data streams. These VC 1, VC 2 channels act as mediators that organize and prioritize packets, ensuring audio receives appropriate handling while data traffic flows through the same infrastructure without direct interference.
3Loss of time
If small buffer sizes are used at network nodes, then latency is reduced, but packet loss increases when buffers overflow during high traffic periods
Solution Approach 1:
The patent performs preliminary buffer allocation based on predicted traffic patterns and pre-established virtual channel priorities. By calculating optimal buffer sizes in advance and reserving bandwidth for audio traffic, the system prevents buffer overflow while maintaining small buffers that minimize latency. The preliminary action ensures packets are directed to appropriate buffers before overflow can occur.
Solution Approach 2:
The patent implements dynamic buffer management where buffer allocation and priority treatment adjust in real-time based on current network conditions. When traffic patterns change, the system dynamically reallocates buffer resources while maintaining the low-latency characteristic through priority-based queuing, preventing overflow without requiring large static buffers.
Data Source
AI summary
Method and Apparatus for delivering audio signals from a source node to a destination node on a network. The apparatus uses a number of switches that transmit prioritized data on a packet network. The switches are coupled to a number of send/receive nodes for sending and receiving digital audio signals on the data network. The audio packet size and the receive buffers are sized to store a minimum possible number of audio samples to minimize latency in processing audio signals arriving at said receive node, but still ensure audio delivery without interruption due to packet data network delay. An additional feature of the invention is recovery of clock synchronization over the same data network by novel arrangement of transmission of timing packets on the network. By sending a multiplicity of packets at irregular intervals a minimum network transit delay can be determined by each of the receive nodes which allows the receive nodes to filter out packet network transit delay error and maintain accurate local clocks.