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The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
S/PDIF is a type of digital audio interconnect used in consumer audio equipment to output audio over relatively short distances. The signal is transmitted over either a coaxial cable with RCA connectors or a fiber optic cable with TOSLINK connectors. S/PDIF interconnects components in home theaters and other digital high-fidelity systems.
AES3 is a standard for the exchange of digital audio signals between professional audio devices. An AES3 signal can carry two channels of PCM audio over several transmission media including balanced lines, unbalanced lines, and optical fiber.
Serial digital interface (SDI) is a family of digital video interfaces first standardized by SMPTE in 1989. For example, ITU-R BT.656 and SMPTE 259M define digital video interfaces used for broadcast-grade video. A related standard, known as high-definition serial digital interface (HD-SDI), is standardized in SMPTE 292M; this provides a nominal data rate of 1.485 Gbit/s.
In digital audio electronics, a word clock or wordclock is a clock signal used to synchronise other devices, such as digital audio tape machines and compact disc players, which interconnect via digital audio signals. Word clock is so named because it clocks each audio sample. Samples are represented in data words.
Clock synchronization is a topic in computer science and engineering that aims to coordinate otherwise independent clocks. Even when initially set accurately, real clocks will differ after some amount of time due to clock drift, caused by clocks counting time at slightly different rates. There are several problems that occur as a result of clock rate differences and several solutions, some being more appropriate than others in certain contexts.
In computer networking and telecommunications, TDM over IP (TDMoIP) is the emulation of time-division multiplexing (TDM) over a packet-switched network (PSN). TDM refers to a T1, E1, T3 or E3 signal, while the PSN is based either on IP or MPLS or on raw Ethernet. A related technology is circuit emulation, which enables transport of TDM traffic over cell-based (ATM) networks.
I²S, is an electrical serial bus interface standard used for connecting digital audio devices together. It is used to communicate PCM audio data between integrated circuits in an electronic device. The I²S bus separates clock and serial data signals, resulting in simpler receivers than those required for asynchronous communications systems that need to recover the clock from the data stream. Alternatively I²S is spelled I2S or IIS. Despite the similar name, I²S is unrelated to the bidirectional I²C (IIC) bus.
AES47 is a standard which describes a method for transporting AES3 professional digital audio streams over Asynchronous Transfer Mode (ATM) networks.
The Precision Time Protocol (PTP) is a protocol used to synchronize clocks throughout a computer network. On a local area network, it achieves clock accuracy in the sub-microsecond range, making it suitable for measurement and control systems. PTP is currently employed to synchronize financial transactions, mobile phone tower transmissions, sub-sea acoustic arrays, and networks that require precise timing but lack access to satellite navigation signals.
Many services running on modern digital telecommunications networks require accurate synchronization for correct operation. For example, if telephone exchanges are not synchronized, then bit slips will occur and degrade performance. Telecommunication networks rely on the use of highly accurate primary reference clocks which are distributed network-wide using synchronization links and synchronization supply units.
SMPTE 292 is a digital video transmission line standard published by the Society of Motion Picture and Television Engineers (SMPTE). This technical standard is usually referred to as HD-SDI; it is part of a family of standards that define a Serial Digital Interface based on a coaxial cable, intended to be used for transport of uncompressed digital video and audio in a television studio environment.
In audio and broadcast engineering, Audio over Ethernet is the use of an Ethernet-based network to distribute real-time digital audio. AoE replaces bulky snake cables or audio-specific installed low-voltage wiring with standard network structured cabling in a facility. AoE provides a reliable backbone for any audio application, such as for large-scale sound reinforcement in stadiums, airports and convention centers, multiple studios or stages.
AES51 is a standard first published by the Audio Engineering Society in June 2006 that specifies a method of carrying ATM cells over Ethernet physical structure intended in particular for use with AES47 to carry AES3 digital audio transport structure. The purpose of this is to provide an open standard, Ethernet based approach to the networking of linear (uncompressed) digital audio with extremely high quality-of-service alongside standard Internet Protocol connections.
Multichannel Audio Digital Interface (MADI) standardized as AES10 by the Audio Engineering Society (AES) defines the data format and electrical characteristics of an interface that carries multiple channels of digital audio. The AES first documented the MADI standard in AES10-1991, and updated it in AES10-2003 and AES10-2008. The MADI standard includes a bit-level description and has features in common with the two-channel AES3 interface.
AES53 is a standard first published in October 2006 by the Audio Engineering Society that specifies how the timing markers specified in AES47 may be used to associate an absolute time-stamp with individual audio samples. AES47 specifies a format for the transmission of digital audio over asynchronous transfer mode (ATM) networks. A recommendation is made to refer these timestamps to the SMPTE epoch which in turn provides a reference to UTC and GPS time. It thus provides a way of aligning streams from disparate sources, including synchronizing audio to video, and also allows the total delay across a network to be controlled when the transit time of individual cells is unknown. This is most effective in systems where the audio is aligned with an absolute time reference such as GPS, but can also be used with a local reference.
Synchronous Ethernet, also referred as SyncE, is an ITU-T standard for computer networking that facilitates the transference of clock signals over the Ethernet physical layer. This signal can then be made traceable to an external clock.
The Open Control Architecture (OCA) is a communications protocol architecture for control, monitoring, and connection management of networked audio and video devices. Such networks are referred to as "media networks".
AES67 is a technical standard for audio over IP and audio over Ethernet (AoE) interoperability. The standard was developed by the Audio Engineering Society and first published in September 2013. It is a layer 3 protocol suite based on existing standards and is designed to allow interoperability between various IP-based audio networking systems such as RAVENNA, Livewire, Q-LAN and Dante.
Audio Video Bridging (AVB) is a common name for the set of technical standards which provide improved synchronization, low-latency, and reliability for switched Ethernet networks. AVB embodies the following technologies and standards: