TL;DR: AES67 brings the convergence of telecom, radio and television studio audio and intercom, and fundamentally enables new workflows, coordinating and combining the efforts of production staff and talent, in geographically combined or diverse locations.
Abstract: Traditionally, due to previous practical technical limitations, the audio quality of telecommunications and intercom systems was not as high as studio audio. Applications requiring long distance high quality audio required the use of specialized provisioning and equipment in parallel to the existing telecommunications systems. — IP computer networks have long since erased the distinction between the local LAN and global networks. As high speed wide area networks (WANs) with better performance and reliability have come online, the reality of erasing the difference of local vs remote audio presents itself. — AES67 is the protocol designed to take advantage of this capability for audio. Using AES67, the audio quality of telecommunications and intercommunications can be the same as in-studio audio, and furthermore, the systems directly interconnected by using the single interoperability protocol. — This shift is more significant than eliminating the economic redundancy of parallel systems. It fundamentally enables new workflows, coordinating and combining the efforts of production staff and talent, in geographically combined or diverse locations. — Audio traffic is no longer just communication; it can be contribution as well. — Combining the ease of making a connection like a phone call, the practically unlimited flexibility of routing of the network, with the pristine high fidelity of digital studio audio, AES67 brings the convergence of telecom, radio and television studio audio and intercom.
TL;DR: This article is based on audio over IP, and then AES67 standards and RAVENNA solutions are introduced to analyze its history and technology advantages.
Abstract: With the continuous development of the digital audio signal transmission technology,digital network transmission has become the industry's most popular and widely accepted one of the means of transport. This article is based on audio over IP,and then AES67 standards and RAVENNA solutions are introduced to analyze its history and technology advantages.
TL;DR: In this paper, the authors provide an overview of AES67 and explore how it can be used with SMPTE ST 2022-6, ST 2110, and Video Services Forum (VSF) TR03 uncompressed video to form a complete solution.
Abstract: Independent audio routing, or audio breakaway, is a standard aspect of today’s TV production workflow and is functionality that will need to be implemented in IP as the industry transitions away from Serial Digital Interface (SDI). Fortunately, the existing AES67 standard for audio over IP meets this objective and eliminates the need for the industry to reinvent the wheel. Not only is there already AES67 equipment deployed in the audio industry, but using this standard also enables significant new workflow opportunities. This paper provides an overview of AES67 and explores how it can be used with SMPTE ST 2022–6, ST 2110, and Video Services Forum (VSF) TR03 uncompressed video to form a complete solution. AES67 uses the IEEE-1588 PTP timing standard and, combining AES67, IEEE-1588, SMPTE ST 2059, and SMPTE ST 2022–6 using the new VSF Technical Recommendation 04 (TR04), provides a solution for maintaining A/V alignment throughout the production workflow.
TL;DR: An overview of AES67 is provided and how it can be used with SMPTE ST 2022–6, ST 2110, and Video Services Forum (VSF) TR03 uncompressed video to form a complete solution for maintaining A/V alignment throughout the production workflow is explored.
Abstract: Independent audio routing, or audio breakaway, is a standard aspect of today’s TV production workflow and is functionality that will need to be implemented in IP as the industry transitions away from Serial Digital Interface (SDI). Fortunately, the existing AES67 standard for audio over IP meets this objective and eliminates the need for the industry to reinvent the wheel. Not only is there already AES67 equipment deployed in the audio industry, but using this standard also enables significant new workflow opportunities. This paper provides an overview of AES67 and explores how it can be used with SMPTE ST 2022–6, ST 2110, and Video Services Forum (VSF) TR03 uncompressed video to form a complete solution. AES67 uses the IEEE-1588 PTP timing standard and, combining AES67, IEEE-1588, SMPTE ST 2059, and SMPTE ST 2022–6 using the new VSF Technical Recommendation 04 (TR04), provides a solution for maintaining A/V alignment throughout the production workflow.
TL;DR: In this article, an AoIP (Audio over IP) technology-based multi-channel audio remote transmission system is presented, which is suitable for broadcast-quality professional occasions having higher requirements for audio quality, such as launching pad transmission links, outfield live rooms, inter-station connection and live broadcast.
Abstract: The present invention discloses an AoIP (Audio over IP) technology-based multi-channel audio remote transmission system. The system comprises an audio transmitting end, a wide area network access end,a wide area network dedicated line, an audio receiving end and a calibration module. Audios transmitted by the AoIP (Audio over IP) technology-based multi-channel audio remote transmission system conform to the GY/T 304-2016 (AES67) format standard at a transmission layer; the audio transmitting end and the audio receiving end implement strict media clock synchronization, usually adopt 48KHz sampling and 24bit quantization, do not use any audio coding compression modes, and can realize original lossless transmission; and the audio transmitting end and the audio receiving end can be configuredwith standard analog line interfaces and AES/EBU digital interfaces according to needs and are suitable for broadcast-quality professional occasions having higher requirements for audio quality, suchas launching pad transmission links, outfield live rooms, inter-station connection and live broadcast. The system is low in latency, is provided with a plurality of channels, supports two-way transmission and can realize audio stream and management data link-sharing simultaneous transmission.