Electronics Guide

AES3 (AES/EBU)

AES3, commonly known as AES/EBU after its sponsoring organizations (Audio Engineering Society and European Broadcasting Union), is the professional standard for two-channel digital audio interconnection. Introduced in 1985 and refined through subsequent revisions, AES3 remains the reference standard for professional audio equipment interconnection.

The AES3 interface transmits two channels of linear PCM audio over a single balanced connection. The data stream is self-clocking, embedding timing information within the signal using biphase mark coding (BMC), where each bit boundary is marked by a transition. This encoding scheme allows receivers to recover both data and clock from a single signal without a separate clock connection.

Standard AES3 uses balanced 110-ohm twisted-pair cable with XLR connectors, typically allowing cable runs of 100 meters or more at standard sample rates. The signal level is nominally 2 to 7 volts peak-to-peak into the specified impedance. Professional equipment universally supports AES3, making it the default choice for connecting high-quality digital audio devices.

The AES3 data format includes 32 bits per audio sample period, organized as a subframe containing 4 synchronization bits, 4 auxiliary bits (sometimes used to extend word length to 24 bits), 20 audio sample bits, and 4 status bits. Two subframes form a frame, and 192 frames form a block. The channel status bits, accumulated over each block, convey important metadata including sample rate, emphasis, copy protection status, and source identification.

AES3 supports sample rates from 32 kHz to 192 kHz, with the data rate scaling proportionally. At 48 kHz, the bit rate is approximately 3.072 Mbps. Higher sample rates require shorter maximum cable lengths due to increased signal bandwidth and associated cable losses.

S/PDIF (IEC 60958 Type II)

S/PDIF, the Sony/Philips Digital Interface Format, is the consumer counterpart to AES3. While sharing the same basic data format and encoding scheme, S/PDIF differs in its physical layer specifications and channel status bit usage, making the two interfaces similar but not identical.

The most common S/PDIF implementation uses 75-ohm coaxial cable with RCA connectors, operating at a lower signal level (0.5 to 1 volt peak-to-peak) than AES3. This configuration is electrically unbalanced and more susceptible to interference, limiting practical cable lengths to a few meters in typical installations. Despite these limitations, coaxial S/PDIF provides reliable performance for connecting consumer and semi-professional equipment.

Optical S/PDIF, using the TOSLINK (Toshiba Link) connector system, transmits the same data format over plastic optical fiber using red LED light sources. Optical transmission eliminates ground loops and provides complete electrical isolation between connected devices. However, the bandwidth limitations of plastic fiber and the relatively slow response of LED transmitters restrict optical S/PDIF to sample rates of 96 kHz or below in most implementations. Cable lengths are typically limited to 5-10 meters.

S/PDIF channel status bits differ from AES3 professional format, reflecting consumer-oriented metadata including copy protection (SCMS - Serial Copy Management System), category codes for device identification, and consumer-format specific information. Equipment designed for professional applications should properly handle both professional and consumer channel status formats.

AES-3id (75-Ohm AES3)

AES-3id adapts the AES3 professional format to 75-ohm unbalanced coaxial cable with BNC connectors, a format common in video facilities. This variant maintains the professional channel status format while using physical connections that integrate seamlessly with broadcast video infrastructure. Signal levels are 1 volt peak-to-peak, higher than consumer S/PDIF but lower than balanced AES3.

The 75-ohm impedance matches standard video cabling, allowing AES-3id signals to share cable infrastructure with video signals and use standard video patch bays. This compatibility makes AES-3id the preferred AES3 variant in television production facilities where digital audio must integrate with extensive video routing systems.

ADAT Lightpipe

ADAT Lightpipe, developed by Alesis for their ADAT digital multitrack recorder in 1991, has become a widely adopted format for transmitting eight channels of digital audio over a single optical fiber. Despite its proprietary origins, ADAT Lightpipe's utility led to its implementation across hundreds of audio products from numerous manufacturers.

The ADAT format transmits eight channels of 24-bit audio at sample rates up to 48 kHz over standard TOSLINK optical connections. The data stream is essentially eight S/PDIF-like subframes multiplexed together with appropriate synchronization. At higher sample rates, the eight channels can be traded for fewer channels at higher resolution: four channels at 96 kHz (S/MUX2) or two channels at 192 kHz (S/MUX4) using sample multiplexing techniques.

ADAT Lightpipe's eight-channel capacity and use of common optical connectors made it enormously popular for expanding the input and output capacity of audio interfaces and mixing consoles. Many audio interfaces include ADAT ports alongside their analog connections, enabling system expansion using external preamp units or converters. The format remains widely supported despite its technical limitations compared to more modern multichannel interfaces.

TDIF (TASCAM Digital Interface)

TDIF was developed by TASCAM for their DA-88 digital multitrack recorder and related products. Like ADAT, TDIF carries eight channels of digital audio, but uses a 25-pin D-sub connector with unbalanced electrical signaling rather than optical transmission.

The electrical nature of TDIF allows for longer cable runs than optical ADAT while maintaining reliable eight-channel transmission. TDIF supports sample rates up to 96 kHz using similar sample multiplexing techniques to ADAT. The format saw widespread use in recording studios during the 1990s and early 2000s, though it has become less common as networked audio protocols have emerged.

TDIF's 25-pin connector also carries word clock and other synchronization signals, simplifying system integration. This self-contained approach made TDIF well-suited for connecting TASCAM tape machines with mixing consoles and other processing equipment.

MADI (AES10)

MADI, the Multichannel Audio Digital Interface, was standardized as AES10 in 1991 and has become the dominant format for high-channel-count point-to-point connections in professional installations. A single MADI link carries up to 64 channels of 24-bit audio at 48 kHz, or 32 channels at 96 kHz, providing massive capacity over a single cable.

MADI supports three physical layer options: 75-ohm coaxial cable with BNC connectors (supporting runs up to 100 meters), multimode fiber optic cable (up to 2 kilometers), and single-mode fiber (up to 10 kilometers). The coaxial format is most common in studio installations, while fiber is preferred for broadcast and live sound applications where distance and electrical isolation are important.

The MADI data stream runs at a fixed 125 Mbps, with each frame containing 64 time slots regardless of the actual channel count in use. Each time slot contains 32 bits: 24 audio bits plus validity, user, channel status, and parity bits. The format supports variable sample rates from 32 kHz to 96 kHz (or 192 kHz with reduced channel count) and can accommodate non-audio data in unused channels.

MADI has proven remarkably durable, with installations from the 1990s still operating reliably. Its high channel count, simple point-to-point topology, and predictable latency make it well-suited for connecting digital mixing consoles with stage boxes, multitrack recorders, and other high-channel-count equipment. Many professional facilities maintain MADI infrastructure alongside newer networked audio systems.

AES50 (SuperMAC)

AES50, originally developed by Sony as SuperMAC, is a high-bandwidth digital audio interface designed for professional live sound applications. A single AES50 link carries 48 bidirectional channels of 24-bit audio at up to 96 kHz over standard CAT5e Ethernet cable with etherCON or standard RJ45 connectors.

Despite using Ethernet physical layer components, AES50 is not a true Ethernet protocol and cannot share network infrastructure with standard IT traffic. The cable runs point-to-point between devices, with a maximum length of 100 meters using standard cables or longer with specialized low-capacitance cables.

AES50's 96-channel bidirectional capacity (48 channels each direction) and low latency make it popular in live sound systems, particularly those based on Behringer/Midas digital mixing consoles that implement AES50 natively. The interface enables remote stage boxes to connect to front-of-house consoles over a single cable, greatly simplifying system wiring while providing ample channel capacity.

USB Audio

USB has become the dominant connection method between computers and audio interfaces, driven by universal availability, adequate bandwidth, and bus power capability. USB audio is standardized through the USB Audio Class specifications, which define how audio devices communicate with host computers using standard USB protocols.

USB Audio Class 1.0, based on USB 1.1, supports sample rates up to 96 kHz and two channels in each direction without additional drivers on most operating systems. While limited in capability, UAC 1.0 devices offer broad compatibility and work reliably across different platforms.

USB Audio Class 2.0, requiring USB 2.0 High Speed (480 Mbps), dramatically expands capability to support sample rates up to 384 kHz, bit depths to 32 bits, and channel counts limited primarily by the audio device's design rather than the interface specification. Native UAC 2.0 support is included in macOS and Linux; Windows requires drivers until version 10 (1703), after which native support is available.

USB Audio Class 3.0 adds support for USB 3.0 SuperSpeed connections and introduces new features including device-side audio processing description and power management improvements. However, adoption has been limited, with most professional audio interfaces continuing to use UAC 2.0.

Many professional USB audio interfaces implement proprietary extensions beyond the standard class specifications, requiring manufacturer-supplied drivers but enabling features such as lower latency, direct monitoring, DSP mixing, and custom sample rate support. The trade-off between driver dependency and expanded capability varies by application.

Thunderbolt Audio

Thunderbolt, originally developed by Intel and Apple, provides extremely high bandwidth (10-40 Gbps depending on version) and low latency PCIe-like connectivity over a single cable. For audio applications, Thunderbolt enables interfaces with very high channel counts and extremely low round-trip latency.

Unlike USB Audio Class, Thunderbolt audio interfaces require manufacturer-specific drivers. The PCIe-like architecture allows audio hardware to appear as a native PCIe device to the host computer, enabling performance comparable to internal expansion cards. Professional interfaces supporting 64 or more simultaneous channels with sub-millisecond latency are available.

Thunderbolt 3 and 4 use the USB-C connector form factor, potentially causing confusion with USB-C audio interfaces. The underlying protocols are completely different, though some devices support both Thunderbolt and USB operation with different performance characteristics. Thunderbolt's higher cost and platform limitations (historically Mac-centric, now more broadly available) have kept it primarily in the professional market segment.

FireWire (IEEE 1394)

FireWire was once the preferred connection for professional computer audio interfaces, offering guaranteed bandwidth, bus power, and peer-to-peer connectivity that USB 1.1 could not match. FireWire 400 (IEEE 1394a) provided 400 Mbps bandwidth, while FireWire 800 (IEEE 1394b) doubled this to 800 Mbps.

FireWire audio interfaces proliferated from the late 1990s through the 2000s, with many professional products supporting dozens of channels over a single connection. The isochronous transfer mode guaranteed timing-critical audio data would receive priority over other traffic, ensuring reliable operation even on busy buses.

FireWire has been deprecated by its creators and is no longer included on new computers, though legacy interfaces remain in use with adapter solutions. USB 2.0 and later standards eventually surpassed FireWire's practical capabilities for audio, while Thunderbolt has addressed the high-performance segment FireWire once served.

Dante

Dante, developed by Audinate, has become the dominant networked audio protocol in professional audio installations. Dante transmits uncompressed digital audio over standard Ethernet networks, supporting hundreds of channels with sample-accurate synchronization across large installations. The protocol is licensed to over 500 manufacturers and implemented in thousands of products.

Dante operates over standard IT network infrastructure using Layer 3 IP protocols, allowing audio traffic to share networks with other data types when properly configured. The system supports 1 Gbps and 100 Mbps Ethernet, with higher bandwidth networks enabling more channels. A single 1 Gbps network can theoretically carry over 500 channels of 48 kHz audio, though practical limits depend on network configuration.

Synchronization in Dante uses IEEE 1588 Precision Time Protocol (PTP), distributing a master clock across the network to all devices. This approach achieves sub-microsecond timing accuracy, ensuring sample-accurate alignment across all connected equipment. One device on the network acts as the PTP grandmaster, with automatic fallback to backup masters if the primary fails.

Dante Controller software provides centralized management of routing, device configuration, and network monitoring. Audio routing is subscription-based: receiving devices subscribe to channels published by transmitting devices, with connections established automatically across the network. This model scales efficiently and simplifies system changes.

Latency in Dante systems is configurable based on network requirements. Default latency settings of 1 ms are typical for dedicated audio networks, while shared networks may require higher latency settings for reliable operation. Device-to-device latency adds additional fixed delays depending on hardware implementation.

AVB (Audio Video Bridging)

AVB is a set of IEEE standards (802.1BA, 802.1Qat, 802.1Qav, and others) that enable guaranteed-bandwidth, low-latency audio and video transmission over Ethernet. Unlike protocols that work over standard networks, AVB requires network switches that implement the AVB standards, ensuring reliable performance through hardware-level quality of service guarantees.

The AVB timing model uses IEEE 802.1AS (a profile of IEEE 1588 PTP) for network synchronization. Stream Reservation Protocol (802.1Qat) allows devices to reserve bandwidth before transmitting, ensuring capacity will be available. Forwarding and Queuing for Time-Sensitive Streams (802.1Qav) manages traffic shaping to maintain guaranteed latency.

Apple's implementation of AVB in macOS and iOS devices brought the technology to consumer products, though professional adoption has been more limited compared to Dante. The requirement for AVB-capable switches, while providing performance guarantees, has been a barrier to adoption in some markets. However, AVB's standards-based approach appeals to organizations concerned about vendor lock-in.

Milan, developed by the Avnu Alliance, is a certification program that ensures AVB interoperability between products from different manufacturers. Milan-certified devices must meet specific performance requirements and pass interoperability testing, addressing early AVB products' compatibility issues.

AES67

AES67 is an AES standard that defines interoperability for high-performance audio-over-IP systems. Rather than creating a new protocol, AES67 specifies a common baseline from existing standards that different audio-over-IP implementations can use for interoperation. This approach allows Dante, Ravenna, Livewire, and other proprietary protocols to exchange audio through AES67 compatibility modes.

AES67 specifies use of RTP (Real-time Transport Protocol) for audio transport, IEEE 1588 PTP for synchronization, and SDP (Session Description Protocol) for session management. The standard defines specific parameter ranges for sample rates (44.1 kHz to 96 kHz), bit depths (16, 24, or 32 bits), and packet timing (125 microseconds to 4 milliseconds per packet).

The significance of AES67 lies in breaking down barriers between different manufacturer ecosystems. A facility with Dante infrastructure can receive audio from a Ravenna-based remote unit if both support AES67 mode. Broadcast facilities increasingly mandate AES67 compliance to ensure flexibility in equipment selection and future-proof their installations.

AES67 does not address discovery and connection management, which vary between implementations. ST 2110-30, part of the SMPTE ST 2110 professional media over IP suite, builds on AES67 while adding broadcast-specific features and standardized discovery mechanisms.

Ravenna

Ravenna, developed by ALC NetworX, is an open technology for real-time audio and media distribution over IP networks. Like Dante, Ravenna enables multichannel audio distribution with low latency and tight synchronization, but with an emphasis on open standards and public specification availability.

Ravenna implements AES67 natively, providing direct interoperability with other AES67-compliant systems. The protocol supports sample rates up to 384 kHz and arbitrary channel counts limited by network bandwidth. Ravenna's open specification has attracted adoption in broadcast infrastructure, where interoperability and long-term availability are critical concerns.

The protocol uses standard networking protocols (RTP, PTP, SDP) and can operate over standard network infrastructure. IGMP multicast allows efficient distribution of audio streams to multiple receivers without duplicating network traffic. Ravenna's architecture supports both dedicated audio networks and appropriately configured shared networks.

Livewire and Livewire+

Livewire, developed by Axia Audio (part of the Telos Alliance), was one of the earliest audio-over-IP systems, introduced in 2003 for broadcast applications. The system enables radio broadcast facilities to distribute audio between studios, transmitters, and production areas over standard Ethernet infrastructure.

Livewire+ (AES67) extends the original protocol with AES67 compatibility, allowing interoperation with other AES67-compliant systems while maintaining backward compatibility with existing Livewire installations. This hybrid approach protects existing infrastructure investments while enabling connectivity with newer equipment.

The Livewire system includes not only audio transport but also comprehensive control capabilities, enabling panels, mixing consoles, and automation systems to communicate over the same network. This integration has made Livewire particularly popular in radio broadcast, where centralized control of distributed audio routing is essential.

Traditional MIDI

MIDI (Musical Instrument Digital Interface), introduced in 1983, is not an audio interface but a control protocol that has become inseparable from digital audio systems. MIDI transmits performance data (note on/off, velocity, controller changes) rather than audio, enabling synthesizers, samplers, and computers to communicate.

Traditional MIDI uses a 31.25 kbaud serial connection with 5-pin DIN connectors. The relatively slow data rate can introduce timing delays when transmitting complex data, a limitation that has driven development of faster alternatives. Despite this limitation, traditional MIDI's simplicity and universal support ensure continued relevance.

MIDI messages are organized into channels (16 per cable) and include note messages, control change messages, program changes, and system messages. The protocol's flexibility has enabled applications far beyond its original musical instrument focus, including lighting control, show automation, and equipment remote control.

USB MIDI and MIDI over USB

USB MIDI class compliance allows MIDI devices to connect to computers without special drivers on most operating systems. The USB connection provides much higher bandwidth than traditional MIDI, eliminating timing bottlenecks for complex data. Most modern MIDI controllers and interfaces use USB as their primary connection to computers.

USB MIDI can carry multiple virtual MIDI cables (up to 16 per device, each with 16 channels) over a single USB connection. This multiplied channel capacity addresses limitations of traditional single-cable MIDI installations. However, USB MIDI typically requires a computer in the signal path, unlike traditional MIDI's direct device-to-device capability.

MIDI 2.0

MIDI 2.0, ratified in 2020, represents the first major update to the MIDI specification since 1983. The new protocol maintains backward compatibility with MIDI 1.0 while adding bidirectional communication, higher resolution (32-bit velocity, controllers, and pitch bend), and property exchange for device configuration.

MIDI 2.0's Universal MIDI Packet (UMP) format provides a new transport container that can carry both MIDI 1.0 and MIDI 2.0 messages. This flexibility allows gradual adoption, with new high-resolution features available when both devices support them while maintaining compatibility with legacy equipment.

Per-note controllers in MIDI 2.0 enable expressive techniques previously impossible with channel-wide controllers. Each note can have independent pitch bend, pressure, and timbre control, supporting multidimensional controllers and polyphonic expression. This capability addresses a long-standing limitation that creative workarounds had only partially solved.

Network MIDI and RTP-MIDI

RTP-MIDI, specified in RFC 6295, enables MIDI transmission over IP networks using the Real-time Transport Protocol. Apple's implementation, built into macOS and iOS, allows MIDI communication between devices over WiFi or wired Ethernet, eliminating cable runs and enabling new workflows.

Network MIDI offers advantages including extended range, the ability to pass through routers and even across the internet (with appropriate latency considerations), and integration with existing network infrastructure. The protocol includes jitter compensation to maintain timing accuracy despite network variations.

Word Clock

Word clock is a dedicated synchronization signal used to lock multiple digital audio devices to a common timing reference. The signal is a square wave at the sample rate frequency (48 kHz, 96 kHz, etc.) distributed over 75-ohm coaxial cable with BNC connectors. One device acts as the master clock, with all other devices configured as slaves.

Proper word clock distribution requires attention to cable lengths, termination, and signal quality. Clock signals should follow a star topology where possible, with individual cables from the master to each slave. Daisy-chaining clock signals can work but introduces additional jitter at each connection. The final device in any chain should include a 75-ohm termination to prevent reflections.

High-quality word clock generators use temperature-compensated or oven-controlled oscillators (TCXO or OCXO) to achieve extremely low jitter. While modern converters include capable internal clocks, external master clocks can provide performance benefits in critical applications, particularly when many devices must synchronize.

Embedded Clock

Most digital audio interfaces (AES3, S/PDIF, ADAT, MADI) include embedded clock signals that receiving devices can extract and lock to. This self-clocking capability simplifies small systems by eliminating the need for separate word clock connections. The receiving device's phase-locked loop (PLL) extracts timing from the incoming data stream.

Embedded clock synchronization works well for point-to-point connections but becomes problematic in systems with multiple sources. If two AES3 inputs both expect to provide clock, one must be selected as master while the other is sample-rate converted or the sources must share a common external clock. Understanding clock routing is essential for system design.

IEEE 1588 Precision Time Protocol

IEEE 1588 PTP provides precision time synchronization over packet networks, enabling the tight timing required for professional audio distribution over Ethernet. PTP messages exchanged between devices allow each node to calculate network delay and synchronize its local clock to a grandmaster clock.

PTP achieves sub-microsecond synchronization accuracy on well-configured networks, far exceeding what is achievable with protocols like NTP. Hardware timestamping in network interfaces improves accuracy by measuring packet timing at the physical layer rather than relying on software timestamps that include processing delays.

Audio-over-IP protocols including Dante, AVB, and AES67 all use PTP (or profiles thereof) for synchronization. The grandmaster clock is automatically selected based on configurable priorities and clock quality metrics, with automatic failover if the grandmaster becomes unavailable. This distributed approach provides resilience appropriate for mission-critical audio systems.

Jitter Reduction Techniques

Jitter in digital audio systems originates from clock source instability, signal transmission effects, and clock recovery circuits. Reducing jitter improves conversion accuracy, as DACs and ADCs produce cleaner output when sampling occurs at precisely regular intervals. Multiple techniques address jitter at different points in the signal chain.

Low-jitter clock sources use high-quality oscillators with excellent short-term stability. Crystal oscillators vary in quality, with better units achieving jitter in the picosecond range. For the most demanding applications, rubidium or GPS-disciplined oscillators provide exceptional stability.

Jitter attenuators and reclocking circuits clean up degraded clock signals by locking a low-jitter local oscillator to the incoming signal's average frequency while rejecting high-frequency timing variations. These circuits are particularly valuable when clock signals have traveled through long cables or multiple distribution stages.

Asynchronous sample rate conversion (ASRC) provides another approach, decoupling the receiving device's conversion clock from the incoming audio stream entirely. The incoming samples are interpolated to the local clock rate, eliminating clock-related jitter (though introducing its own conversion artifacts that are typically minimal in high-quality implementations).

Sample Rate Conversion

Sample rate conversion changes audio between different sampling frequencies, essential when connecting equipment operating at different rates or when preparing content for different distribution formats. High-quality sample rate conversion uses sophisticated interpolation algorithms to minimize artifacts.

Synchronous sample rate conversion handles integer ratio conversions (such as 48 kHz to 96 kHz) using relatively simple interpolation. Asynchronous conversion handles arbitrary ratios and can compensate for slight clock frequency differences between devices. Modern ASRC implementations achieve transparency that makes the process inaudible in critical listening.

Sample rate converters are available as standalone hardware units, built into audio interfaces and digital mixers, and as software plugins. Real-time conversion is common in broadcast and live applications, while non-real-time software conversion may use more computationally intensive algorithms for archival or mastering applications.

Format and Protocol Conversion

Format converters translate between different digital audio interface types, enabling equipment with incompatible connections to work together. Common conversions include AES3 to S/PDIF (primarily a channel status and level adaptation), MADI to Dante (requiring buffering and clock domain crossing), and various combinations of legacy and modern interfaces.

Protocol converters for networked audio enable facilities to bridge between different ecosystems. Dante-to-AES67 conversion allows standard Dante networks to connect with AES67-compliant equipment from other manufacturers. These bridges must handle not only audio data but also synchronization, often requiring careful configuration to maintain timing integrity across the conversion.

Embedding and de-embedding convert between audio-only formats and audio carried within video streams (SDI or HDMI). De-embedders extract audio channels from video signals for processing in audio-focused equipment, while embedders insert processed audio back into video streams. These conversions are essential in broadcast and post-production facilities where audio and video must remain synchronized.

Application Requirements

Selecting appropriate digital audio interfaces requires matching technical capabilities with application needs. Recording studios typically prioritize audio quality and channel count, making converter quality and interface capacity primary concerns. Live sound emphasizes reliability, ease of setup, and sufficient channel count for the production scale.

Broadcast applications demand high reliability, redundancy options, and compatibility with video infrastructure. Installation sound focuses on long-term reliability, ease of maintenance, and integration with building systems. Each application context emphasizes different interface characteristics.

Scalability and Future-Proofing

System designs should consider future expansion and technology evolution. Networked audio protocols offer inherent scalability advantages, as adding channels often requires only additional endpoint devices rather than infrastructure changes. Point-to-point interfaces may require additional cable runs and patch points for expansion.

Interoperability standards like AES67 provide some protection against vendor lock-in, ensuring that equipment from different manufacturers can work together. However, proprietary features often exceed standard capabilities, requiring trade-offs between openness and functionality.

Latency Requirements

Different applications tolerate different amounts of latency. Live monitoring requires latency below approximately 10 ms to avoid performer disorientation. Broadcast facilities often work with specific frame-based latencies for lip-sync alignment. Recording and playback applications may tolerate higher latencies that are compensated in software.

Interface technologies vary significantly in achievable latency. Direct point-to-point connections (AES3, MADI) add minimal fixed delay. USB and Thunderbolt latency depends on buffer settings and driver implementation. Networked protocols have configurable latency settings that trade lower latency against increased sensitivity to network conditions.

Infrastructure Integration

The choice of digital audio interfaces affects broader infrastructure decisions. Dedicated audio formats (AES3, MADI) require purpose-built cabling and routing systems. USB and Thunderbolt interfaces depend on computer systems that require their own infrastructure. Networked audio can leverage existing Ethernet infrastructure but may require network upgrades or dedicated audio networks depending on requirements.

Facility design increasingly considers networked audio as the primary distribution method, with traditional interfaces serving specific legacy, redundancy, or specialized roles. However, the installed base of traditional interface equipment ensures continued relevance of point-to-point connections for many years.