70V / 100V — Constant-Voltage Distributed Audio
70V (and 100V) distributed audio systems allow a single amplifier to drive many speakers over long cable runs without impedance-matching calculations, using transformer-coupled taps at each speaker.
AEC — Acoustic Echo Cancellation
DSP algorithm that removes room reflections and far-end voice echo from microphone signals to enable full-duplex conferencing.
AES — Audio Engineering Society and Digital Audio Standards
The Audio Engineering Society (AES) and its key standards — AES3, AES67, MADI, and OCA — that define professional digital audio.
AES67 — Audio over IP Interoperability Standard
AES67 is the SMPTE/AES standard for interoperable audio over IP, defining RTP transport, PTP clocking, and SDP/SAP discovery so that Dante, Ravenna, Livewire, and Q-LAN devices can exchange audio across platforms.
ARC and eARC — Audio Return Channel
ARC and eARC carry audio from a display back to an audio device over the HDMI cable — eliminating a separate audio cable from the display to the DSP or amplifier.
Beamforming — Steered Microphone Arrays
Beamforming uses an array of microphone capsules with DSP spatial filtering to steer a directional pickup pattern toward active speakers, rejecting off-axis noise — the technology behind MXA, Parlé, TeamConnect, and Stem ceiling microphones.
CMRR — Common Mode Rejection Ratio
CMRR quantifies how well a balanced audio input rejects noise and interference that appears equally on both conductors — the key specification that makes balanced audio essential for long cable runs in AV systems.
Codec — Compression/Decompression
A codec encodes and decodes audio or video signals using compression algorithms — understanding codecs determines the quality, latency, and bandwidth trade-offs in video conferencing, AV-over-IP, and streaming systems.
Dante — Digital Audio Network Transport
Industry-standard audio networking protocol by Audinate carrying uncompressed multi-channel digital audio over standard Gigabit Ethernet with sub-millisecond latency.
dB — Decibel Reference Suffixes (dBu, dBFS, dBSPL, dBm, dBV)
Decibel suffixes define the reference point for audio level measurements — confusing dBu with dBFS or dBSPL is one of the most common errors in gain structure setup.
Gain Structure — Setting Levels Through the Signal Chain
Gain structure is the systematic setting of amplification and attenuation at every stage in an audio signal chain to maximize signal-to-noise ratio while maintaining adequate headroom before clipping.
Impedance — Audio and Video Impedance Matching
Impedance determines how audio and video signals are loaded and transferred between devices — mismatched impedance causes signal loss, frequency response errors, and reflections in video systems.
Latency and Jitter — AV Network Timing
Latency is the delay from source to destination; jitter is the variation in that delay — both critically affect live audio, video conferencing, and AV-over-IP systems.
LPCM — Linear Pulse Code Modulation and Audio Formats
LPCM is the uncompressed digital audio standard carried over HDMI and AES3 — understanding LPCM vs. compressed formats (Dolby Digital, DTS, Atmos) determines what a system can de-embed, process, and distribute.
PTP — Precision Time Protocol (IEEE 1588)
PTP (IEEE 1588) synchronizes clocks across a network to sub-microsecond accuracy — essential for Dante, AES67, AV-over-IP, and any system where audio or video devices must share a common time reference.
RT60 — Reverberation Time
RT60 is the time in seconds for a sound to decay 60 dB after the source stops — the primary acoustic measurement that determines whether a room is suited for speech, music, or neither.
RTP and RTSP — Real-Time Transport Protocols
RTP carries real-time audio and video over IP networks; RTSP controls the playback of those streams — together they underpin Dante, AES67, IP camera streaming, and IPTV in AV systems.
SIP — Session Initiation Protocol
SIP is the signaling protocol for VoIP and video conferencing — understanding SIP is essential for integrating telephone interfaces, paging systems, and SIP-based video conferencing endpoints with AV systems.
SNR — Signal-to-Noise Ratio
SNR quantifies how much stronger the desired signal is than the noise floor — a fundamental specification for microphones, preamps, DSPs, and amplifiers that determines how quiet a system can be before background noise becomes audible.
STI — Speech Transmission Index
STI is the standard objective measure of speech intelligibility in a room — a value from 0 to 1 that predicts how clearly spoken words will be understood by listeners, accounting for reverberation, noise, and system distortion.
THD — Total Harmonic Distortion
THD measures the nonlinear distortion added by audio equipment — amplifiers, DSPs, and converters — expressed as a percentage or dB below the fundamental, indicating how cleanly the device reproduces a signal.