The Video Technology Glossary

12G-SDI is a high-speed Serial Digital Interface standard capable of transmitting uncompressed 4K video at up to 60 frames per second over a single coaxial cable. Operating at approximately 12 Gbit/s, it significantly reduces cabling complexity for 4K workflows compared to earlier SDI versions.

3G-SDI is an advancement in the SDI standard, providing a single 2.970 Gbit/s serial link that supports 1080p resolution at up to 60 frames per second. It effectively doubles the data rate of HD-SDI, allowing for higher bitrate and resolution video transmission over a single coaxial cable.

3GP is a multimedia container format defined by the 3rd Generation Partnership Project (3GPP) for 3G UMTS multimedia services. It is designed to reduce storage and bandwidth requirements, making it suitable for mobile phones and low-bandwidth networks.

5.1 surround sound is a common multi-channel audio configuration that uses five full-bandwidth channels and one low-frequency effects (LFE) channel. The five main channels typically consist of a center, front left, front right, surround left, and surround right speaker, providing a foundational immersive audio experience for home theaters and broadcast.

6G-SDI is a Serial Digital Interface standard that supports a data rate of 5.94 Gbit/s, designed to carry 4K video at up to 30 frames per second over a single coaxial cable. It is standardized under SMPTE ST 2081-1 and represents a further evolution in SDI capabilities for higher resolutions.

7.1 surround sound is an enhanced multi-channel audio configuration that builds upon 5.1 by adding two additional discrete audio channels, typically side surround speakers. This setup provides a more enveloping and precise sound field, further enhancing the immersive experience in home theater systems and advanced audio setups.

8-VSB is the radio frequency modulation technique used by ATSC 1.0 for over-the-air digital television transmission in North America. It transmits data using eight discrete amplitude levels on a single sideband carrier, achieving a payload of approximately 19.39 Mbit/s in a 6 MHz channel. 8-VSB was chosen for its spectral efficiency in the North American channel plan, though it is more susceptible to multipath interference than the OFDM modulation used by DVB-T2 and ATSC 3.0.

AAC is a lossy digital audio compression standard designed to provide higher quality audio at lower bitrates compared to its predecessor, MP3. It is widely used for streaming and in various devices, offering efficient compression while maintaining good sound fidelity.

AAC-LC, or Low Complexity, is the basic and most widely supported profile of the Advanced Audio Coding (AAC) standard. It offers efficient audio compression with relatively low computational complexity, making it suitable for a broad range of consumer electronics and general audio applications.

Adaptive Bit Rate (ABR) is a streaming technique that delivers video content by dynamically adjusting the video quality and bitrate based on the user's network conditions and device capabilities. It involves encoding the video into multiple renditions (different bitrates and resolutions) and switching between them seamlessly during playback.

An ABR (Adaptive Bitrate) algorithm is a sophisticated logic implemented within a video player that determines which video rendition (bitrate) to request from the server based on real-time network conditions, buffer levels, and device capabilities. The goal of an ABR algorithm is to maximize video quality while minimizing rebuffering and startup delays, ensuring a smooth and continuous playback experience. These algorithms are crucial for adaptive streaming.

An ABR (Adaptive Bitrate) ladder is a predefined set of video encoding profiles, each specifying a unique combination of resolution and bitrate, created from a single source file. Streaming platforms use this ladder to dynamically switch between different quality levels in real-time based on the viewer's available network bandwidth and device capabilities.

AC-3, commonly known as Dolby Digital, is a lossy audio compression technology developed by Dolby Laboratories for efficient encoding of surround sound. It supports up to 5.1 discrete audio channels and is widely used in home theater systems, DVDs, and broadcast television.

Ancillary data refers to supplementary non-video information embedded within a digital video signal, using the same transport mechanism as the video itself. This data can be carried in either the horizontal (HANC) or vertical (VANC) blanking intervals and includes elements like embedded audio, timecode, and closed captioning.

Anycast routing is a network addressing and routing method where a single IP address is shared by multiple servers, typically CDN edge nodes, in different geographical locations. When a user requests content, network routers direct the request to the nearest available server advertising that IP address, optimizing for proximity and reducing latency.

AS-02 is a media master packaging standard developed by AMWA. It is similar to IMF but utilizes metadata stored in MXF files to describe the different versions derived from underlying essence components. An AS-02 package consists of multiple file items organized in a standardized directory structure.

AS-11 is a family of AMWA (Advanced Media Workflow Association) specifications that define constrained media file formats based on MXF. These specifications are used for the delivery of finished media assets to broadcasters and publishers, ensuring interoperability and compliance with regional delivery requirements.

A comprehensive record generated during broadcast playout that documents all events, programs, commercials, and other media elements that were actually aired, including their exact start and end times. This log is crucial for verification, billing, regulatory compliance, and demonstrating to advertisers when their content was shown.

ASF, or Advanced Systems Format, is Microsoft's proprietary digital audio/digital video container format, primarily for streaming media. WMV (Windows Media Video) is a video codec often encapsulated within the ASF container, along with WMA (Windows Media Audio).

Audio delay refers to a temporal offset between the audio and video components of a media presentation, where the sound is heard either before or after the corresponding visual action. This delay can be intentional, for correction purposes, or unintentional, caused by processing differences in various equipment, leading to lip sync issues.

Audio Description (AD) is an additional narration track that verbally describes key visual elements in media content, such as actions, gestures, scene changes, and on-screen text. This service is designed to make video content accessible to individuals who are blind or have low vision, providing crucial contextual information during natural pauses in dialogue.

Audio loudness refers to the subjective perception of the strength or intensity of a sound, which is influenced by factors beyond just peak amplitude, such as frequency content and duration. It is a psychoacoustic phenomenon that modern audio standards aim to measure and control to ensure a consistent and comfortable listening experience.

Audio sync, or audio synchronization, is the precise alignment of audio tracks with their corresponding visual elements in media production. It ensures that sound and picture play back simultaneously and coherently, preventing distracting discrepancies like lip sync errors and maintaining the integrity of the audiovisual experience.

Audio track multiplexing is the process of combining multiple audio streams, along with video and other data, into a single container file or transmission stream. This technique allows for the delivery of various audio options, such as different languages or descriptive audio, within a single media asset, enabling viewers to select their preferred audio experience.

Audio watermarking is the process of embedding a hidden, inaudible piece of information (a watermark) directly into an audio signal. This digital signature can be used for various purposes, including copyright protection, content tracking, and broadcast monitoring, allowing for the identification of the audio source or owner without affecting the perceived audio quality.

AOMedia Video 1 (AV1) is an open, royalty-free video coding format developed by the Alliance for Open Media (AOMedia) as a successor to VP9. It is designed for internet video delivery, offering significant compression efficiency improvements over previous codecs like VP9 and H.264.

AVI, or Audio Video Interleave, is a proprietary multimedia container format introduced by Microsoft in 1992 as part of Video for Windows. It can contain both audio and video data in an uncompressed file container, allowing synchronous audio-with-video playback.

AVOD, or Advertising Video On Demand, is a monetization model where users can access video content for free, with the service provider generating revenue by displaying advertisements. This model is distinct from subscription-based services and pay-per-view, relying entirely on ad impressions to support content delivery. Many free streaming platforms utilize an AVOD model.

AVPlayer is a class within Apple's AVFoundation framework, providing the core functionality for playing local and remote file-based media, as well as audiovisual media served using HTTP Live Streaming (HLS) on iOS, macOS, tvOS, and watchOS platforms. It offers precise control over media playback, timing, and synchronization, making it the fundamental component for video and audio playback in Apple's ecosystem.

AVS3 is China's third-generation audio and video coding standard, designed for ultra-high-definition (UHD) video, virtual reality (VR), and other advanced multimedia applications. It aims to provide superior compression efficiency compared to H.265/HEVC, supporting 8K resolution, HDR, and wide color gamut.

A scheduled interruption in a video stream during which one or more advertisements are played. Ad breaks are signalled in live streams using SCTE 35 cue messages or in VOD manifests via timeline markers, and their duration and frequency are governed by the content owner's monetisation policy.

The real-time process by which an ad server or supply-side platform selects which advertisement to serve into a given ad slot, based on audience data, targeting parameters, floor prices, and available demand. Decisioning typically occurs within milliseconds to meet the latency requirements of live SSAI workflows.

A time-coded signal embedded in a media manifest or elementary stream that indicates the start and/or end of an ad opportunity. In HLS, ad markers appear as EXT-X-CUE-OUT and EXT-X-CUE-IN tags; in DASH, they are represented as EventStream elements. Ad markers are derived from SCTE 35 splice messages.

A group of advertisements played sequentially within a single ad break. An ad pod may contain pre-roll, mid-roll, or post-roll slots and is assembled by the ad decisioning system based on available inventory, targeting, and competitive separation rules.

A platform that stores, manages, and delivers advertising creative to end users. In streaming contexts, the ad server receives ad requests from the SSAI stitcher (containing viewer and context metadata), performs decisioning, and returns a VAST or VMAP response containing the selected ad creative URLs and tracking beacons.

The server-side process of seamlessly splicing advertisement segments into a content stream so that the resulting manifest and media segments appear as a single, uninterrupted stream to the player. Ad stitching handles codec alignment, bitrate matching, and segment boundary synchronisation to avoid visible splice artefacts.

The capability to deliver different advertisements to different viewers watching the same content simultaneously, based on household or individual-level targeting data. Addressable advertising is enabled by SSAI systems that personalise the ad slate per session rather than broadcasting a single national ad feed.

The use of demographic, behavioural, contextual, or geographic data to select advertisements most relevant to a specific viewer or household. In OTT and streaming, targeting data is passed to the ad server via query-string parameters in the ad request URL, often including device type, content genre, geographic region, and consent signals.

ATSC is the international, non-profit organisation that develops voluntary standards for digital television and advanced media. Its standards govern over-the-air digital broadcast television in North America and several other countries, defining the technical specifications for transmission, audio, video, and data services. The ATSC standards family includes ATSC 1.0 (the original digital TV standard), ATSC 2.0 (enhanced interactivity), and the next-generation ATSC 3.0 (NextGen TV).

The original ATSC digital television standard, deployed in North America from the late 1990s. ATSC 1.0 uses 8-VSB (8-level Vestigial Sideband) modulation and MPEG-2 video compression, replacing the NTSC analogue system. It supports resolutions up to 1080i and 720p HD, and remains the baseline over-the-air broadcast standard in the United States, Canada, Mexico, and South Korea.

ATSC 3.0, commercially branded as NextGen TV, is the most significant advancement in over-the-air broadcast television since the transition to digital. Unlike its predecessors, ATSC 3.0 is an IP-based standard that delivers broadcast signals using OFDM modulation, enabling 4K UHD video (HEVC), HDR (HDR10, Dolby Vision), immersive audio (AC-4, MPEG-H), and broadband internet integration. Key capabilities include targeted advertising, emergency alerting, datacasting, and the ability to combine broadcast and broadband signals (broadcast-broadband convergence). It is not backwards-compatible with ATSC 1.0 receivers and requires a new tuner or set-top box.

One of the most commercially significant features of ATSC 3.0 / NextGen TV, enabling broadcasters to deliver household- or device-level personalised advertisements over the air — a capability previously exclusive to OTT and streaming platforms. By combining the broadcast signal with broadband return-path data, ATSC 3.0 allows dynamic ad insertion at the receiver level, opening new programmatic revenue streams for over-the-air broadcasters.

AC-4 is Dolby's next-generation audio coding standard, designed for broadcast and streaming applications requiring immersive audio, efficient compression, and personalisation. It is one of the mandatory audio codecs in ATSC 3.0 / NextGen TV and supports object-based audio (enabling immersive experiences like Dolby Atmos), dialogue enhancement, accessibility features, and efficient delivery of multiple audio services within a single stream.

A B-frame, or Bi-directionally predicted frame, offers the highest compression efficiency by predicting its content from both preceding and succeeding I-frames or P-frames. B-frames require both past and future reference frames for decoding, making them more complex but highly efficient for reducing file size.

BGP, or Border Gateway Protocol, is a standardized exterior gateway protocol that exchanges routing and reachability information between autonomous systems (AS) on the internet. CDNs leverage BGP to announce their IP prefixes, allowing them to influence traffic routing and direct user requests to the most optimal edge PoP, thereby enhancing performance and reliability.

Bit depth, also known as color depth, refers to the number of bits used to represent the color information for each pixel in a video frame. Higher bit depths (e.g., 10-bit or 12-bit) allow for a greater range of colors and more subtle gradations between tones, reducing banding artifacts and improving overall image fidelity compared to lower bit depths (e.g., 8-bit).

Bitrate refers to the amount of data, typically measured in kilobits per second (kbps) or megabits per second (Mbps), that is processed or transmitted per unit of time in a video stream. A higher bitrate generally results in better video quality but also requires more bandwidth and storage, while a lower bitrate reduces file size and bandwidth consumption at the expense of quality.

Bitrate switching is the dynamic adjustment of the video stream's quality (bitrate) in real-time based on the viewer's network conditions and device capabilities. This process is fundamental to adaptive bitrate (ABR) streaming, ensuring that the player can seamlessly switch between different video renditions to maintain continuous playback with the best possible quality. It prevents buffering and optimizes the viewing experience.

BOLA, or Buffer Occupancy-based Lyapunov Algorithm, is a specific type of buffer-based ABR algorithm designed to achieve near-optimal video quality while minimizing rebuffering. It uses Lyapunov optimization techniques to make bitrate decisions by balancing the desire for higher quality with the need to maintain a stable buffer. BOLA is known for its theoretical guarantees and practical performance in adaptive streaming.

In broadcasting, branding encompasses the strategies and visual elements used by television or radio channels to establish a distinct identity, differentiate themselves, and foster viewer loyalty. This includes on-air graphics, logos, jingles, and overall presentation that collectively define the channel's image and appeal to its target audience.

BT.2020, also known as Rec. 2020 or ITU-R Recommendation BT.2020, is a wide color gamut standard for Ultra High Definition (UHD) television. It defines a significantly larger color space than BT.709, enabling the display of a much broader range of colors, which is crucial for HDR content and delivers a more vivid and realistic visual experience.

BT.2100, also known as Rec. 2100 or ITU-R Recommendation BT.2100, is a comprehensive international standard for High Dynamic Range (HDR) television. It defines image parameters for HDR production and international exchange, encompassing both the Perceptual Quantizer (PQ) and Hybrid Log-Gamma (HLG) transfer functions, along with wide color gamut specifications like BT.2020.

BT.709, also known as Rec. 709 or ITU-R Recommendation BT.709, is the international standard for High-Definition (HD) television. It defines the color space, color primaries, white point, and transfer function for HD video, serving as the foundational color standard for most SDR content and displays worldwide.

Buffer-based ABR (Adaptive Bitrate) algorithms prioritize maintaining a healthy video buffer level to prevent rebuffering. These algorithms adjust the video bitrate based on the current buffer occupancy, increasing the bitrate when the buffer is full and decreasing it when the buffer is low. This proactive approach aims to smooth out playback and reduce interruptions, even during network instability.

Byte-range requests are an HTTP/1.1 feature that allows a client to request only a specific portion or range of bytes from a file on a server, rather than downloading the entire resource. This mechanism is crucial for efficient streaming, enabling features like seeking within a video, resuming interrupted downloads, and adaptive bitrate streaming by fetching only necessary media segments.

An HTTP request fired by the player or SSAI server to a tracking URL at specific points during ad playback — typically impression (ad start), first quartile (25%), midpoint (50%), third quartile (75%), and complete (100%). Beacons are defined in the VAST Tracking element and are used by advertisers and agencies to measure ad delivery and viewability.

A technical and commercial strategy that combines traditional broadcast delivery (over-the-air, satellite, or cable) with broadband internet connectivity to create a unified, enriched viewer experience. ATSC 3.0 and HbbTV are the two primary standards enabling broadcast-broadband convergence, allowing broadcasters to supplement linear broadcast signals with on-demand content, personalised advertising, interactive applications, and enhanced metadata delivered over IP.

CABAC is an advanced form of entropy coding used in modern video compression standards like H.264/AVC and HEVC. It achieves high compression efficiency by adaptively modeling the probability distributions of syntax elements based on their context and then using arithmetic coding to represent them with fewer bits, resulting in superior lossless compression.

A cache hit occurs when a user requests content and it is found and served directly from a CDN's edge server cache, resulting in fast delivery. Conversely, a cache miss happens when the requested content is not available in the edge cache, forcing the request to be forwarded to the origin server, which increases latency.

Cache invalidation is the process of marking cached content as stale or expired, prompting the CDN to re-validate or re-fetch the content from the origin server the next time it is requested. Unlike a purge, which removes content, invalidation often just flags it, allowing for more granular control over content freshness without immediate deletion.

Cache purging is the explicit and immediate removal of specific content or all content from a CDN's cache, regardless of its TTL setting. This action is typically performed to ensure that outdated or incorrect content is no longer served to users, forcing edge nodes to fetch the latest version from the origin server upon subsequent requests.

CAMBI is a no-reference banding artifact detector developed by Netflix that quantifies the visibility of banding (contouring) artifacts in video. It operates by extracting pixel-level maps at multiple scales and combining them into a single index, taking into account human contrast sensitivity. CAMBI provides a score that correlates highly with human subjective perception of banding, even when traditional metrics like PSNR and VMAF do not.

CAVLC is a form of entropy coding used in video compression standards like H.264/AVC, serving as a simpler and less computationally intensive alternative to CABAC. It achieves lossless compression by assigning variable-length codes to quantized transform coefficients, with the code lengths adapted based on the context of previously encoded coefficients.

CBCS (Cipher Block Chaining Scheme) is an encryption mode used within the Common Encryption (CENC) standard, particularly for protecting streaming media. It encrypts content using AES in Cipher Block Chaining mode with subsample encryption, meaning only portions of each media sample are encrypted. This method is commonly employed by DRM systems like FairPlay and PlayReady for efficient and secure content delivery.

Constant Bit Rate (CBR) is a rate control method where the encoder attempts to maintain a consistent bitrate throughout the entire video stream. While ensuring predictable bandwidth usage, CBR can lead to fluctuating quality, as complex scenes may be under-encoded and simple scenes over-encoded to maintain the constant rate.

A Content Decryption Module (CDM) is a secure software or hardware component within a client device (e.g., web browser, set-top box) responsible for decrypting Digital Rights Management (DRM) protected content and enforcing usage policies. It interacts with Encrypted Media Extensions (EME) in web browsers to handle license requests, key management, and content decryption in a trusted environment, preventing unauthorized access to the unencrypted media.

CDN analytics refers to the collection, processing, and analysis of data related to content delivery performance, user behavior, and network traffic within a Content Delivery Network. These analytics provide insights into metrics such as cache hit ratios, latency, throughput, geographic distribution of users, and potential security threats, enabling optimization of content delivery strategies.

CDN architecture refers to the distributed network design of a Content Delivery Network, comprising strategically located servers (PoPs) that work together to deliver content efficiently to end-users. This architecture aims to minimize latency and improve content availability by caching data closer to the user, thereby offloading traffic from origin servers.

CDN failover is a mechanism within a multi-CDN strategy that automatically redirects user traffic from a primary CDN to a secondary CDN when the primary experiences an outage, performance issues, or becomes unavailable. This ensures uninterrupted content delivery and maintains a high level of service availability for end-users.

A CDN for live streaming is specifically optimized to deliver real-time video content to a large, geographically dispersed audience with minimal latency and buffering. These CDNs employ specialized protocols and techniques, such as adaptive bitrate streaming and optimized edge caching, to ensure a smooth and high-quality viewing experience for live events.

A CDN for Video on Demand (VOD) is designed to efficiently deliver pre-recorded video content to users upon request. It leverages caching at edge locations to store popular video assets closer to viewers, reducing load times and improving playback quality for on-demand consumption.

CDN latency refers to the delay experienced when a user requests content from a CDN until the first byte of that content is received. It is a critical performance metric influenced by factors such as geographical distance to the edge server, network congestion, and server processing time, with lower latency indicating faster content delivery.

CDN load balancing is the process of distributing incoming user requests across multiple edge servers or even multiple CDN providers to optimize resource utilization, maximize throughput, minimize response time, and prevent overload of any single server. This ensures efficient and reliable content delivery, enhancing the overall user experience.

CDN manifest manipulation involves dynamically altering the manifest files (e.g., HLS or DASH manifests) at the CDN edge to customize the streaming experience for individual users or devices. This can include inserting advertisements, enforcing geo-restrictions, applying digital rights management (DRM) policies, or adapting content based on network conditions.

CDN peering refers to direct interconnection agreements between a CDN provider and other network entities, such as Internet Service Providers (ISPs) or other CDNs. These agreements facilitate the direct exchange of traffic, bypassing intermediate networks, which reduces latency, improves throughput, and lowers transit costs for content delivery.

CDN prefetching is a technique where a CDN proactively fetches and stores content in its edge caches before a user explicitly requests it, based on anticipated user behavior or predefined rules. This aims to reduce latency and improve perceived load times by ensuring content is already available at the edge when a user eventually requests it.

CDN shielding, often referred to as origin shield, is an additional caching layer positioned between a CDN's edge servers and the origin server. Its primary purpose is to protect the origin server from being overwhelmed by numerous cache-fill requests from edge nodes, especially during cache misses or content updates, thereby reducing origin load and improving overall performance.

CDN throughput measures the amount of data successfully delivered by a CDN to end-users over a specific period, typically expressed in bits per second. It indicates the capacity and efficiency of the CDN in handling and delivering content, with higher throughput signifying faster and more robust content delivery capabilities.

CDN tokenisation, or token authentication, is a security mechanism that generates unique, time-sensitive tokens or credentials to validate access requests for content delivered via a CDN. This ensures that only authorized users or applications can access specific content, preventing unauthorized hotlinking or content scraping.

CDN-level Digital Rights Management (DRM) refers to the enforcement of content protection policies directly within the Content Delivery Network infrastructure. This allows for secure delivery of premium content by encrypting media and managing decryption keys at the CDN edge, ensuring that only authorized users with valid licenses can access and play the content.

Common Encryption (CENC) is an ISO standard that defines common encryption and key mapping methods for protecting multimedia content, allowing a single encrypted file to be decrypted by multiple Digital Rights Management (DRM) systems like Widevine, FairPlay, and PlayReady. This standardization simplifies content preparation and distribution by enabling interoperability across different DRM platforms.

A device or software used in broadcasting to create and display static or animated text and graphics, such as lower thirds, credits, and scores, that are keyed into a video stream. CGs are essential for adding informational and branding elements to live and pre-recorded television productions.

An all-in-one integrated playout solution for broadcast television that combines multiple functions like playout, graphics, ingest, and scheduling into a single, often PC-based, system. CiaB simplifies broadcast infrastructure, reduces costs, and provides a flexible solution for originating and managing television channels.

Chroma subsampling is a video compression technique that reduces the amount of color information (chroma) in a video signal while retaining full brightness information (luma), exploiting the human eye's greater sensitivity to luminance than chrominance. Common formats include 4:2:0 (half vertical and horizontal chroma resolution), 4:2:2 (half horizontal chroma resolution), and 4:4:4 (full chroma resolution, no subsampling).

Chunked CMAF refers to the use of Common Media Application Format (CMAF) in conjunction with chunked encoding and chunked transfer encoding to achieve low-latency streaming. In this approach, individual media segments are further subdivided into smaller chunks, allowing for immediate transfer and playback as each chunk is encoded, significantly reducing end-to-end latency.

Chunked transfer encoding is an HTTP/1.1 mechanism that allows a server to send data in a series of self-contained chunks without needing to know the total size of the response body beforehand. Each chunk is preceded by its size, enabling the client to process data incrementally as it arrives, which is crucial for dynamic content and low-latency streaming.

CineForm is a wavelet-based intermediate video codec, originally designed for digital intermediate workflows and later acquired by GoPro. It supports various bit depths and chroma subsampling formats, including 10-bit 4:2:2 and 12-bit 4:4:4, and is known for its constant-quality behavior and stability under iterative recompression, standardized as SMPTE VC-5.

ClearKey is a basic content protection mechanism often used with MPEG-DASH and HLS, where encryption keys are exchanged directly and in the clear (unencrypted) over a secure channel like HTTPS. Unlike full-fledged DRM systems, ClearKey provides a simpler form of encryption without complex license management, making it suitable for content that requires a minimal level of protection or for testing purposes.

Textual representations of the audio portion of a television program, designed for viewers who are deaf or hard of hearing, including dialogue and non-speech elements. CEA-608 is the analog standard for NTSC television, while CEA-708 is the digital standard for ATSC broadcasts, offering enhanced features like improved display options and multiple caption streams.

A Closed GOP is a Group of Pictures where all frames within the GOP can be decoded using only other frames from within that same GOP. This structure ensures that no frame in a closed GOP references frames from a previous GOP, which is beneficial for editing and seamless segment switching in streaming.

CMAF is a container format and set of standards for packaging and delivering HTTP-based media, jointly proposed by Apple and Microsoft. It unifies HLS and MPEG-DASH under a single fMP4 container, reducing duplicate encoding and storage costs. CMAF also employs chunked encoding and chunked transfer encoding to lower latency.

A colour space is a defined range of colors that a video system can capture, reproduce, or display. It acts as a mathematical model that specifies how colors are represented, including their hue, saturation, and brightness. Different colour spaces, such as BT.709 for SDR and BT.2020 for HDR, determine the breadth and accuracy of the colors visible in a video.

A Conditional Access System (CAS) is a security technology primarily used in traditional broadcast television (satellite, cable, terrestrial) to control subscriber access to encrypted premium content. It ensures that only authorized viewers who have paid for a service can decrypt and watch specific channels or programs. CAS typically involves a combination of encryption, scrambling, and subscriber management, often utilizing smartcards to grant or revoke access.

A content key, also known as a decryption key, is the cryptographic key used to decrypt encrypted digital media content within a Digital Rights Management (DRM) system. This key is securely delivered to an authorized client device by a DRM license server after successful authentication and authorization, enabling the playback of protected video or audio.

Constant Quantization Parameter (CQP) is a rate control method where a fixed quantization parameter (QP) is applied to all frames or blocks within a video. This ensures a consistent level of detail preservation, but the resulting bitrate can vary significantly depending on the complexity of the video content.

Constant Rate Factor (CRF) is a quality-based rate control method where the encoder aims to achieve a consistent perceptual quality throughout the video by adjusting the bitrate as needed. A lower CRF value generally results in higher quality and larger file sizes, while a higher CRF value yields lower quality and smaller files.

CTR (Counter) mode is an encryption mode for block ciphers like AES, which effectively turns them into stream ciphers. In the context of video streaming and DRM, CTR mode encrypts data by XORing the plaintext with a keystream generated by encrypting a sequence of counter values. This mode is favored for its ability to allow parallel processing and random access to encrypted data, making it suitable for adaptive streaming where different parts of a video may need to be decrypted independently.

An ad insertion model in which the video player on the client device is responsible for fetching, stitching, and playing advertisements. The player pauses the content stream, requests an ad from the ad server via a VAST call, plays the ad creative, and then resumes content. CSAI is susceptible to ad blockers and can produce visible buffering at ad transitions.

Digital Audio Broadcasting is a digital radio standard used in many countries, particularly in Europe, for broadcasting digital audio services. It offers improved sound quality, more stations, and enhanced data services compared to traditional analog FM radio, with DAB+ being an improved version that uses more efficient audio coding.

dash.js is an open-source JavaScript library that provides a reference client implementation for playing MPEG-DASH (Dynamic Adaptive Streaming over HTTP) content in web browsers. It utilizes MediaSource Extensions (MSE) and Encrypted Media Extensions (EME) to enable adaptive bitrate streaming and DRM-protected playback of DASH streams. dash.js is a foundational tool for developers building DASH-compliant video players for the web.

DCP, or Digital Cinema Package, is a collection of digital files used to store and convey digital cinema (DC) audio, image, and data streams. It is the standard format for distributing movies to digital cinemas, ensuring interoperability and security.

The Discrete Cosine Transform (DCT) is a mathematical transform used in video compression to convert spatial domain pixel data into the frequency domain. This transformation concentrates most of the image information into a few low-frequency coefficients, making it easier to quantize and discard less perceptually important high-frequency information, thus enabling efficient compression.

Deinterlacing is the process of converting interlaced video into a progressive scan format. This technique reconstructs full frames from the alternating fields of an interlaced signal, eliminating artifacts like 'combing' or 'mouse teeth' that can appear on progressive displays, thereby improving video quality for modern screens and digital distribution.

Dirac is an advanced royalty-free video compression format developed by the BBC Research & Development, designed for high-quality video applications across various resolutions and bitrates. It utilizes wavelet-based compression and offers both intra-frame and inter-frame coding, supporting progressive and interlaced content.

Avid DNxHD (Digital Nonlinear Extensible High Definition) is a lossy high-definition video post-production codec developed by Avid, standardized as SMPTE VC-3. It is designed for multi-generation compositing with reduced storage and bandwidth, functioning as an intermediate format for editing and presentation, typically stored in MXF or QuickTime containers.

Avid DNxHR (Digital Nonlinear Extensible High Resolution) is a high-resolution mezzanine and post-production codec developed by Avid, superseding DNxHD to support resolutions beyond 1080p, including 2K, 4K, and 8K. It is an intra-frame only codec, offering multiple compression levels and color codings for flexible workflows, and is not intended for end-user distribution.

Dolby AC-4 is a next-generation audio codec developed by Dolby Laboratories, designed for broadcast and streaming services. It offers high-quality audio at lower bitrates than previous codecs and supports advanced features like immersive audio and personalized listening experiences, including descriptive audio for accessibility.

Dolby Atmos is an immersive audio technology that expands on traditional surround sound systems by adding height channels and object-based audio. It allows sound designers to precisely place and move individual sounds in a three-dimensional space, creating a highly realistic and enveloping listening experience.

Dolby Vision is a proprietary High Dynamic Range (HDR) technology developed by Dolby Laboratories that supports up to 12-bit color depth and uses dynamic metadata to optimize picture quality on a scene-by-scene or frame-by-frame basis. This allows for precise adjustments to brightness, contrast, and color, delivering a superior and more consistent visual experience across a wide range of compatible displays.

Downscaling is the process of reducing the resolution of a video or image to a lower resolution. This is commonly done to decrease file size, reduce bandwidth requirements for streaming, or to make content compatible with lower-resolution display devices. While it saves resources, downscaling inevitably results in a loss of detail and sharpness compared to the original higher-resolution content.

Digital Rights Management (DRM) refers to a set of access control technologies used by content creators and copyright holders to control the use, modification, and distribution of copyrighted digital content, such as video, music, and e-books. Its primary goal is to prevent unauthorized access, copying, and piracy, ensuring that only authorized users can consume the media according to specified rules.

DRM key rotation is a security practice in Digital Rights Management (DRM) systems where the encryption keys used to protect content are regularly changed or updated. This process enhances security by limiting the exposure time of any single key, making it more difficult for unauthorized parties to decrypt and access protected media, especially in long-duration live streams or frequently accessed on-demand content.

A DRM license server is a critical component in a Digital Rights Management system responsible for issuing decryption keys and usage rules (licenses) to client devices requesting access to protected content. When a user attempts to play DRM-protected media, the client application communicates with the license server to obtain a valid license, which then enables the content to be decrypted and played according to the content owner's specified rights.

A DRM license token is a digital credential issued by a DRM license server to a client device, granting permission to access and play specific protected content under defined usage rules. This token typically contains the decryption key for the content and specifies parameters such as playback duration, device limitations, and output restrictions, ensuring that content consumption adheres to the content owner's rights.

DTS, originally Digital Theater Systems, is a family of audio coding technologies known for delivering discrete multichannel sound. It is a lossy compression format that provides high-quality audio for commercial theaters, AV systems, and home entertainment, often competing with Dolby Digital.

DTS-HD refers to a suite of audio codecs from DTS, including DTS-HD Master Audio and DTS-HD High Resolution Audio. DTS-HD Master Audio is a lossless codec that delivers bit-for-bit identical audio to the studio master, while DTS-HD High Resolution Audio is a lossy, high-quality format. Both support multi-channel surround sound.

DTS:X is an object-based immersive audio codec that allows sound elements to be positioned and moved freely in a three-dimensional space. Unlike channel-based systems, DTS:X adapts to various speaker layouts, providing a flexible and lifelike audio experience for home theaters and other entertainment environments.

DVB-CSA (Digital Video Broadcasting - Common Scrambling Algorithm) is the encryption algorithm specified by the DVB Project for scrambling video streams in digital television broadcasting. It is a core component of Conditional Access Systems (CAS), used to protect pay-TV content by ensuring that only authorized set-top boxes or integrated televisions can decrypt and display the broadcast signals. Several versions exist, with DVB-CSA2 and DVB-CSA3 offering enhanced security over earlier iterations.

A technology that replaces or inserts advertisements into video streams dynamically at the time of delivery, rather than encoding them into the content at ingest. DAI encompasses both server-side (SSAI) and client-side (CSAI) approaches and enables personalised, targeted advertising in live and VOD streams.

A programmatic advertising platform used by advertisers and agencies to purchase ad inventory across multiple publishers and exchanges through automated, real-time bidding. In streaming ad tech, DSPs connect to SSPs via OpenRTB to bid on available ad impressions in OTT and CTV environments.

DVB is a suite of internationally accepted open standards for digital television, developed and maintained by the DVB Project — a consortium of over 170 companies. DVB standards define the technical specifications for digital broadcast transmission across satellite (DVB-S, DVB-S2, DVB-S2X), cable (DVB-C, DVB-C2), terrestrial (DVB-T, DVB-T2), and handheld (DVB-H) delivery. DVB is the dominant broadcast standard in Europe, Australia, and much of Asia, Africa, and South America.

DVB-T2 is the second-generation DVB terrestrial broadcast standard, offering significantly higher capacity and robustness than its predecessor DVB-T. Using advanced modulation techniques (OFDM with up to 256-QAM) and LDPC/BCH forward error correction, DVB-T2 can carry HD and UHD channels alongside interactive HbbTV services. It is the current terrestrial broadcast standard in the UK (Freeview HD), Germany, Italy, Sweden, and many other countries.

DVB-I is a DVB standard that defines how broadcast-quality linear TV services can be delivered over the public internet, without requiring a traditional broadcast tuner. It specifies a service discovery and selection framework that allows a DVB-I client (on a Smart TV, set-top box, or mobile device) to discover and play linear channels delivered via IP, including those with HbbTV applications. DVB-I is designed to extend the reach of broadcast services to devices and markets without broadcast reception infrastructure.

E-AC-3, or Dolby Digital Plus, is an enhanced audio compression scheme based on the AC-3 codec, offering increased bitrates and support for more audio channels, up to 7.1. It provides higher quality and more immersive surround sound experiences, commonly used in streaming services and advanced home theater setups.

Edge nodes, also known as edge servers, are computing devices located at the periphery of a CDN's network, geographically close to end-users. These nodes store cached content and serve as the primary point of interaction for user requests, significantly reducing the physical distance data must travel and improving delivery speed.

A table within DVB Service Information (SI) that contains data concerning events or programs, such as event names, start times, durations, and content descriptions. The EIT is crucial for populating Electronic Programme Guides (EPGs), allowing viewers to see what is currently airing and what is scheduled for the future.

Encrypted Media Extensions (EME) is a W3C specification that provides a JavaScript API for web browsers to interact with Content Decryption Modules (CDMs). EME enables web applications to play back protected audio and video content by allowing the browser to select content protection mechanisms, control license/key exchange, and execute custom license management algorithms, without exposing the decryption keys directly to the web application.

An encoder is a hardware device or software application that converts raw video and audio signals into a compressed digital format. This compression is crucial for efficient storage and transmission over networks, making the media suitable for streaming. Encoders prepare content for further processing, such as packaging and distribution.

An encoding ladder is the collection of different video renditions—varying in resolution, bitrate, and sometimes codec—that are generated during the transcoding process for adaptive streaming. It serves as the blueprint for delivering optimal video quality across diverse devices and network conditions, ranging from low-bandwidth mobile connections to high-speed 4K displays.

Encrypted Media Extensions (EME) is a W3C API that enables web applications to interact with content protection systems, allowing for the playback of encrypted and Digital Rights Management (DRM)-protected media. EME provides a communication channel between the browser's media engine and a Content Decryption Module (CDM), facilitating secure content delivery and preventing unauthorized access. It is essential for premium video streaming services.

End-to-end latency represents the total time taken for a video signal or data packet to travel from its source to its final destination, encompassing all processing, transmission, and rendering delays along the entire delivery chain. In streaming, it's the delay from the point of content origination (e.g., live event capture) to its display on the viewer's device. Minimizing end-to-end latency is crucial for interactive and real-time applications.

Entitlement management, in the context of broadcast and streaming video, refers to the process of defining, assigning, and enforcing user permissions and access rights to digital content and services. It ensures that only subscribers with valid entitlements can access specific content, features, or service tiers, playing a crucial role in subscription management, content monetization, and compliance with licensing agreements.

Entropy coding is a lossless data compression technique applied after quantization in video compression. It reduces redundancy by assigning shorter binary codes to frequently occurring symbols (e.g., quantized coefficients) and longer codes to less frequent ones, thereby minimizing the overall bitrate without any loss of information.

A menu-based system that provides users of television and other media with a schedule of current and upcoming programs, often with detailed information about each program. EPGs serve as a digital TV guide, allowing viewers to navigate channels, set reminders, and access program descriptions.

Essential Video Coding (EVC) is a video compression standard developed by MPEG, designed to offer improved compression efficiency over AVC and HEVC with a clear, royalty-free baseline profile. It aims to provide a balance between performance and licensing costs, targeting a wide range of applications from mobile to UHD broadcasting.

ExoPlayer is an open-source, application-level media player for Android developed by Google, offering an alternative to Android's built-in MediaPlayer API. It provides advanced features for adaptive streaming (DASH, HLS), customizability, and support for various media formats and DRM schemes. ExoPlayer is widely adopted for building robust and flexible video playback experiences on Android devices.

An HLS manifest tag that marks the end of an ad break and the resumption of content. It is paired with EXT-X-CUE-OUT and is inserted by the SSAI system or packager when the SCTE 35 splice_insert() command with out_of_network_indicator=0 is received.

An HLS manifest tag that signals the start of an ad opportunity, typically carrying a DURATION attribute indicating the expected length of the ad break in seconds. It is generated from a SCTE 35 splice_insert() command with out_of_network_indicator=1 and is used by SSAI systems to trigger ad decisioning.

F4V is a Flash Video container format based on the ISO Base Media File Format, similar to MP4. It was introduced by Adobe to address some limitations of the original FLV format, offering better support for H.264 video and AAC audio.

FairPlay is a Digital Rights Management (DRM) technology developed by Apple Inc. for protecting digital content, primarily streaming media delivered via HTTP Live Streaming (HLS) to Apple devices such as iOS, macOS, and Apple TV. It ensures secure delivery and playback by encrypting content and managing decryption keys, preventing unauthorized access and distribution within the Apple ecosystem.

FAST, or Free Ad-Supported Streaming TV, is a streaming model that offers linear, channel-based video content to viewers at no subscription cost. Revenue is generated through advertisements embedded within the programming, mimicking the traditional television broadcast model. FAST services provide a wide range of content, often curated into themed channels, accessible on various internet-connected devices.

FFmpeg is a free and open-source software project consisting of a suite of libraries and programs for handling video, audio, and other multimedia files. It is a powerful command-line tool capable of decoding, encoding, transcoding, muxing, demuxing, streaming, filtering, and playing virtually any multimedia format, making it an essential tool in broadcast and streaming workflows.

In the context of video and audio, fingerprinting is a technology that generates a unique digital signature or "fingerprint" for a piece of content. This fingerprint, a compact mathematical representation of the content's characteristics, allows for rapid identification and comparison of media files, even if they have been modified, transcoded, or re-recorded. It is primarily used for content identification, copyright enforcement, and tracking unauthorized usage across various platforms.

FLAC, or Free Lossless Audio Codec, is an audio coding format for lossless compression of digital audio. Unlike lossy codecs, FLAC compresses audio without discarding any information, meaning the decompressed audio is bit-for-bit identical to the original source, making it popular for archiving and high-fidelity audio playback.

FLV, or Flash Video, is a container file format used to deliver video over the Internet using Adobe Flash Player. It can encapsulate video streams using codecs such as Sorenson Spark or On2 VP6, and audio streams using MP3 or AAC.

Fragmented MP4 is a version of the standard MP4 file format that logically partitions media into 'moof-mdat' pairs (fragments). This structure allows for immediate playback after downloading a small initial segment and facilitates adaptive streaming by enabling dynamic adjustment of video quality based on network conditions.

Forensic watermarking is a content protection technique that embeds a unique, imperceptible identifier into each individual video or audio stream delivered to a user. If pirated content is discovered, this embedded watermark can be extracted and analyzed to trace the source of the leak, thereby deterring unauthorized distribution and aiding in the identification of infringers.

Frame rate, commonly expressed in frames per second (FPS), is the frequency at which consecutive images (frames) are captured or displayed in a video. A higher frame rate results in smoother motion and more detailed representation of fast-moving objects, while a lower frame rate can appear choppy but may be sufficient for less dynamic content or to conserve bandwidth.

The percentage of available ad slots that are successfully filled with paid advertising. A fill rate below 100% results in house ads, slates, or content being played in unfilled slots. Fill rate is influenced by demand availability, targeting strictness, floor prices, and the breadth of demand sources connected to the ad decisioning system.

A rule that limits the number of times a specific advertisement is shown to the same viewer within a defined time window. Frequency capping prevents ad fatigue, improves viewer experience, and is enforced by the ad server using viewer identity signals such as device ID, cookie, or IP address.

Geo-restriction, also known as geo-blocking, is a CDN feature that allows content providers to restrict or grant access to content based on the geographical location of the end-user. This is typically achieved by identifying the user's IP address and comparing it against a database of geographical locations, enabling compliance with licensing agreements or regional content rights.

Glass-to-glass latency measures the total time delay from the moment light enters the lens of a camera (the 'first glass') to when the processed image is displayed on a viewer's screen (the 'second glass'). This comprehensive metric accounts for all stages of the video pipeline, including capture, encoding, transmission, decoding, and rendering. It is a critical indicator for real-time applications where end-to-end responsiveness is essential.

A Group of Pictures (GOP) is a sequence of video frames that defines the order in which I-frames, P-frames, and B-frames appear within a compressed video stream. The GOP structure significantly impacts compression efficiency, random access capabilities, and editing flexibility.

GOP structure refers to the arrangement and sequence of I-frames, P-frames, and B-frames within a Group of Pictures. This structure dictates how frames are encoded and decoded, influencing factors like compression ratio, latency, and the ability to seek or edit video content.

A software and/or hardware system responsible for generating and rendering visual elements, such as text, logos, animations, and virtual sets, for broadcast. These engines are crucial for creating dynamic on-screen graphics that enhance viewer engagement and provide essential information during live broadcasts.

GXF, or General Exchange Format, is a file format designed for the exchange of broadcast television material. It is a wrapper that can contain various video, audio, and data essences, facilitating interoperability between different broadcast systems.

Advanced Video Coding (AVC), also known as H.264, is a widely adopted video compression standard that can encode high-quality video at lower bit rates compared to older standards. It is compatible with most streaming protocols and container formats, and is extensively used in Blu-ray and various streaming services.

High Efficiency Video Coding (HEVC), also known as H.265, is a video compression standard that succeeds H.264/AVC, offering improved compression efficiency and better picture quality. It is capable of supporting higher resolutions, including 8K, and is used by many video hosting servers.

Versatile Video Coding (VVC), also known as H.266, is the latest video compression standard developed as the successor to HEVC, aiming for approximately 50% better compression efficiency. It supports a broad range of applications, including 8K and higher resolutions, HDR, and 360-degree video.

Hardware encoding utilizes dedicated circuitry within a device, such as a GPU or a specialized chip, to perform video compression. This method offers significant speed advantages and reduces CPU load, making it ideal for real-time applications like live streaming and gaming, though it may sometimes yield slightly lower quality or less flexibility compared to software encoding.

High-Definition Serial Digital Interface (HD-SDI) is a standard within the SDI family that specifically supports the transmission of high-definition uncompressed digital video signals. It operates at a data rate of 1.485 Gbit/s over coaxial cable, enabling high-quality video transfer in professional broadcast environments.

High Dynamic Range (HDR) video technology significantly expands the range of luminance and color that can be displayed, offering brighter whites, deeper blacks, and a wider, more vibrant color palette compared to standard dynamic range (SDR). This results in a more immersive and realistic visual experience, bringing images closer to what the human eye perceives in the real world.

HDR10 is an open standard for High Dynamic Range (HDR) video, utilizing a 10-bit color depth, the Rec. 2020 color space, and the PQ (Perceptual Quantizer) transfer function. It uses static metadata, meaning the display settings are applied uniformly across the entire video content, providing a significant improvement in contrast and color vibrancy over SDR.

HDR10+ is an advanced High Dynamic Range (HDR) video standard that builds upon HDR10 by incorporating dynamic metadata. This dynamic metadata allows for scene-by-scene or even frame-by-frame adjustments to brightness, contrast, and color, optimizing the HDR presentation for each specific scene and display device, resulting in a more accurate and impactful visual experience than static HDR10.

HDS is an adaptive bitrate streaming method developed by Adobe, delivering MP4 video content over HTTP connections. It was primarily designed for use with Adobe Flash Player and Adobe AIR, and it allows for both on-demand and live streaming with content caching.

High-Efficiency Advanced Audio Coding (HE-AAC), also known as AAC+, builds upon AAC-LC by incorporating Spectral Band Replication (SBR) technology. This enhancement significantly improves compression efficiency, delivering near CD-quality audio at very low bitrates, making it ideal for mobile and streaming applications.

HE-AAC v2 further enhances HE-AAC by adding Parametric Stereo (PS) technology to Spectral Band Replication (SBR). This combination allows for even greater compression efficiency, particularly for stereo signals at extremely low bitrates, making it highly suitable for digital radio and mobile streaming services.

HESP, or High Efficiency Streaming Protocol, is an HTTP-based adaptive bitrate streaming protocol designed to achieve ultra-low latency and fast channel changes for live video. Developed by THEO Technologies, HESP aims to deliver sub-second latency at scale with reduced bandwidth requirements, making it suitable for interactive live events and real-time applications. It represents an evolution in streaming protocols for highly responsive experiences.

HESP is a new video streaming protocol designed for ultra-low latency and fast channel changes over HTTP networks. It aims to deliver high-quality video streaming at scale with significantly reduced bandwidth requirements, addressing limitations of traditional HTTP Adaptive Streaming protocols.

Hybrid Log-Gamma (HLG) is an HDR transfer function jointly developed by the BBC and NHK, primarily designed for broadcast television. HLG is backward-compatible with SDR displays, meaning SDR TVs can display an HLG signal without special conversion, albeit without the full HDR benefits. It uses a hybrid curve that combines a gamma curve for the lower luminance range and a logarithmic curve for the higher luminance range.

HLS is an HTTP-based adaptive bitrate streaming communications protocol developed by Apple Inc. It works by breaking the overall stream into a sequence of small HTTP-based file downloads, each downloading one short chunk of an overall potentially unbounded transport stream. A list of available streams, encoded at different bit rates, is sent to the client using an extended M3U playlist.

HLS encryption (AES-128) refers to the use of the Advanced Encryption Standard (AES) with a 128-bit key in Cipher Block Chaining (CBC) mode to encrypt entire media segments within an HTTP Live Streaming (HLS) manifest. This method provides a basic level of content protection by scrambling the video and audio data, requiring a decryption key to be obtained by the player to enable playback. It is a common encryption method for HLS streams, offering a balance between security and compatibility.

HLS ingest refers to the process of feeding live video streams encoded in the HLS (HTTP Live Streaming) format into a streaming platform or server. While HLS is primarily a delivery protocol, HLS ingest allows for direct contribution of HLS-formatted content, though it typically has higher latency than RTMP or WebRTC ingest due to its segment-based nature.

hls.js is an open-source JavaScript library that implements an HTTP Live Streaming (HLS) client in web browsers that do not natively support HLS. It relies on HTML5 video and MediaSource Extensions (MSE) to parse and play HLS streams, enabling adaptive bitrate playback of HLS content across a broad range of web platforms. It is a crucial component for HLS compatibility in non-Apple environments.

HTTP/2 push is a feature of the HTTP/2 protocol that allows a server, including a CDN edge server, to proactively send resources to a client before the client explicitly requests them. This can significantly improve page load times by reducing the number of round trips required to fetch critical assets like CSS, JavaScript, and images.

HTTP/3 is the third major version of the Hypertext Transfer Protocol, which defines how information is exchanged on the World Wide Web. It is built on top of the QUIC transport protocol, leveraging QUIC's features like reduced connection establishment time, improved congestion control, and elimination of head-of-line blocking to enhance speed, security, and reliability.

Huffman coding is a widely used entropy coding technique that achieves lossless data compression by assigning variable-length codes to input symbols based on their frequency of occurrence. More frequent symbols receive shorter codes, while less frequent symbols receive longer codes, resulting in an overall reduction in data size.

Hybrid monetisation in streaming refers to the strategy of combining multiple revenue models, such as subscription (SVOD), advertising (AVOD), and transactional (TVOD/PVOD), within a single platform or service. This approach aims to maximize revenue by catering to diverse consumer preferences and offering flexible access options. It allows platforms to diversify income streams and attract a broader audience.

HbbTV is an open European standard that combines digital broadcast television with broadband internet services on a single connected TV platform. Developed by the HbbTV Association — a consortium of broadcasters, consumer electronics manufacturers, and technology companies — HbbTV enables broadcasters to deliver interactive applications, catch-up TV, EPGs, targeted advertising, and data services alongside the broadcast signal. HbbTV is the dominant hybrid broadcast standard in Europe and has been adopted in over 50 countries. The standard is built on web technologies (HTML5, CSS, JavaScript) and uses the broadcast signal to trigger and deliver applications to the viewer's TV.

The widely deployed first major version of the HbbTV standard, based on CE-HTML and the Open IPTV Forum (OIPF) specifications. HbbTV 1.4 introduced the core hybrid broadcast-broadband model, enabling broadcasters to launch interactive red-button applications, catch-up services, and enhanced EPGs on connected televisions. It is supported by the majority of Smart TVs sold in Europe from 2012 onwards.

A major evolution of the HbbTV standard introducing HTML5-based application delivery, adaptive bitrate streaming (MPEG-DASH), HEVC support, and advanced features including targeted advertising, multi-screen synchronisation, and companion screen interaction. HbbTV 2.0 aligns the broadcast-broadband hybrid platform with modern web standards, enabling broadcasters to deliver richer, more personalised experiences on connected TVs.

The ongoing series of incremental updates to HbbTV 2.0, including versions 2.0.2, 2.0.3, and later revisions. HbbTV 2.x updates have progressively added support for UHD and HDR content, improved targeted advertising capabilities (including server-side ad insertion), enhanced accessibility features, and tighter integration with DVB-I (Internet-delivered broadcast services). HbbTV 2.x is the current baseline for new Smart TV deployments in Europe.

A feature of HbbTV 2.0 and later that enables broadcasters to replace or supplement linear broadcast advertisements with personalised, targeted ads delivered over broadband. Using a combination of broadcast scheduling signals and IP-delivered ad creatives, HbbTV targeted advertising allows broadcasters to compete with OTT platforms for programmatic ad revenue while retaining the scale and reach of linear broadcast. It is a key commercial driver for HbbTV adoption across European broadcasters.

An I-frame, or Intra-coded frame, is a complete video frame that contains all the necessary visual information for decoding without reference to any other frames. While offering the least compression, I-frames are crucial for random access, seeking, and error recovery in video streams.

An IDR frame, or Instantaneous Decoder Refresh frame, is a special type of I-frame that signals to the decoder that no subsequent frames will reference any frames prior to the IDR frame. This effectively resets the decoding process, ensuring that all frames following an IDR can be decoded independently, which is critical for stream switching and error resilience.

IMF, or Interoperable Master Format, is a series of SMPTE standards for the exchange of finished, high-quality audio-visual masters between facilities and for long-term archiving. It is component-based, allowing for efficient versioning and localization without re-encoding the entire master.

Immersive audio is a broad term for technologies and techniques that create a three-dimensional soundscape, placing the listener within the audio environment. It goes beyond traditional stereo or surround sound by incorporating height and depth, often utilizing object-based audio and specialized speaker configurations to deliver a highly realistic and engaging auditory experience.

An initialization segment is a crucial component in adaptive bitrate streaming protocols, particularly in MPEG-DASH and HLS (when using fMP4). It contains essential metadata about the media stream, such as codec information, track details, and resolution, which is required by the player to initialize the media decoder before it can process subsequent media segments.

Inter-frame compression is a video coding technique that reduces temporal redundancy by encoding only the differences between successive frames, rather than compressing each frame entirely. This method exploits the high correlation between frames in a video sequence to achieve significantly higher compression ratios than intra-frame coding.

Interlaced scanning (denoted by 'i', e.g., 1080i) displays video by drawing alternating lines of each frame in two passes, first odd lines then even lines, which was common in older broadcast television to conserve bandwidth. Progressive scanning (denoted by 'p', e.g., 1080p) draws all lines of each frame sequentially in a single pass, resulting in a smoother, more stable image with less motion blur, and is the standard for modern displays and digital video.

Intra-frame compression is a video coding technique that compresses each video frame independently, utilizing spatial redundancy within that single frame to reduce file size and bitrate. This method does not rely on information from other frames for decoding.

Inverse telecine (IVTC) is a video processing technique used to reverse the telecine process, restoring film-originated content that has been converted to video back to its original progressive 24 frames per second. This process removes the redundant fields and artifacts introduced by telecine, resulting in a cleaner, more film-like progressive video output.

IPTV, or Internet Protocol Television, is a system that delivers television content using Internet Protocol (IP) networks, rather than traditional terrestrial, satellite, or cable formats. While similar to OTT in using IP, IPTV is often managed by telecommunication providers who offer it as part of a bundled service, ensuring a dedicated network for quality of service. It typically provides scheduled programming and on-demand content.

IPTV middleware is a software platform that acts as an intermediary between the IPTV service provider's backend systems and the end-user's viewing devices (e.g., set-top boxes, smart TVs, mobile apps). It manages and delivers various IPTV services, including live TV, video-on-demand, and interactive features, while providing the graphical user interface (GUI) and handling functions like content management, subscriber authentication, and billing integration.

Integrated Services Digital Broadcasting is a Japanese broadcasting standard for digital television and digital radio, designed to supersede analog systems like NTSC-J and MUSE. It supports various services including satellite (ISDB-S), terrestrial (ISDB-T), and cable (ISDB-C) broadcasting, and is known for its advanced features like interactive television and mobile reception (1seg).

ISOBMFF is a container file format that defines a general structure for files containing time-based multimedia data like video and audio. It is designed to be flexible, extensible, and independent of any particular network protocol, serving as the basis for many other media file formats such as MP4 and 3GP.

An industry trade organisation that develops and maintains technical standards for digital advertising, including VAST, VMAP, VPAID, SIMID, and OMID. IAB standards define how ad creatives are delivered, rendered, and measured across web, mobile, and connected TV environments.

A single instance of an advertisement being served and rendered to a viewer. In streaming ad tech, an impression is recorded when the ad creative begins playing and the impression beacon fires. Impression counting methodology varies by standard — VAST defines impression as the moment the first frame of the ad is displayed.

JPEG 2000 (JP2) is an image compression standard and coding system based on discrete wavelet transforms, developed to supersede the original JPEG standard. It offers both lossless and lossy compression, supports multiple resolutions, progressive transmission, and is used in applications like digital cinema and medical imagery.

JPEG XS (ISO/IEC 21122) is an image and video codec designed for low-complexity and low-latency implementations, offering visually and mathematically lossless quality. It is primarily used for professional video over IP, real-time video storage, and sensor compression, prioritizing quality and low latency over maximum compression efficiency.

A Key ID (KID) is a globally unique identifier associated with a specific content key used in Digital Rights Management (DRM) systems. When a media player encounters encrypted content, it uses the KID embedded in the content's metadata to request the corresponding decryption key from a DRM license server. This allows the server to identify and deliver the correct key for content decryption.

Key rotation is a security practice in Digital Rights Management (DRM) where the cryptographic keys used to encrypt content are regularly changed or updated. This process minimizes the risk of a compromised key being used for unauthorized access over an extended period, enhancing the overall security of the content. It is particularly important for live streaming and long-form video-on-demand content.

Last-mile delivery, in the context of CDNs, refers to the final segment of the content delivery path, from the nearest CDN edge server to the end-user's device. This phase is critical for user experience as it often involves local network conditions and can significantly impact perceived latency and content loading times.

Low Complexity Enhancement Video Coding (LCEVC) is an MPEG standard that enhances existing video codecs by adding a low-complexity enhancement layer. It improves compression efficiency and picture quality, particularly at lower bitrates, by encoding a base layer with any codec (e.g., AVC, HEVC) and then applying a small, computationally efficient enhancement layer.

Linear streaming refers to the delivery of video content over the internet in a scheduled, channel-based format, similar to traditional broadcast television. Viewers watch programs at specific times on specific channels, without the ability to pause, rewind, or fast-forward beyond the live point. This contrasts with on-demand viewing, where content can be accessed at any time.

Lip sync, short for lip synchronization, is the technical process of matching a person's lip movements with spoken dialogue or sung lyrics. In broadcast and streaming, maintaining accurate lip sync is critical for a natural and immersive viewing experience, as even slight discrepancies can be highly noticeable and distracting to the audience.

A live streaming workflow encompasses the entire process of capturing, encoding, processing, distributing, and playing back real-time video content to viewers. This workflow involves components like cameras, encoders, media servers, CDNs, and player applications, all working in concert to deliver content with minimal delay. It is designed for events that occur and are consumed simultaneously.

Lossless compression is a data compression method that allows the original video data to be perfectly reconstructed from the compressed data without any loss of information. While it results in larger file sizes compared to lossy methods, it is crucial for applications where data integrity and perfect fidelity are paramount, such as archival or professional editing workflows.

Lossy compression is a data compression method that achieves significant file size reduction by permanently discarding some of the original video data, typically information deemed less perceptually important to the human eye. This technique is widely used in broadcast and streaming to optimize bandwidth and storage, accepting a controlled reduction in quality for much smaller file sizes.

Loudness normalisation is the process of adjusting the perceived loudness of audio programs to a consistent target level, rather than just peak levels. Standards like EBU R128 (Europe) and ATSC A/85 (North America) define methodologies and target loudness values (e.g., -23 LUFS for EBU R128, -24 LUFS for ATSC A/85) to ensure a uniform listening experience across different content and channels.

Low-Latency DASH is an extension of the MPEG-DASH protocol designed to reduce end-to-end streaming delay, enabling near real-time video delivery. It achieves lower latency by utilizing smaller media segments (chunks) and efficient manifest updates, allowing for quicker adaptation to network changes and minimizing buffering.

Low-Latency HLS is an extension of the HTTP Live Streaming (HLS) protocol developed by Apple, designed to significantly reduce end-to-end latency in live video streaming. It achieves this by introducing partial media segments and using HTTP/2 push, allowing for sub-two-second latencies while maintaining the scalability and reliability of traditional HLS.

LPCM, or Linear Pulse Code Modulation, is a specific type of Pulse Code Modulation where the quantization levels are linearly uniform. It represents uncompressed digital audio data, providing a direct digital representation of the original analog waveform without any data loss or compression artifacts, commonly found on Blu-ray discs and professional audio equipment.

LRA, or Loudness Range, is a metric that quantifies the dynamic variation of loudness within an audio program over time. It measures the difference between the softest and loudest sections of a program, providing insight into its dynamic characteristics and helping to ensure that content maintains an appropriate dynamic spread for its intended distribution platform.

An encoding of SMPTE timecode data in an audio signal, commonly recorded on a VTR track or other storage media. It is distributed as an audio signal and can be read when the recording is in motion, but is ineffective when the recording is stationary.

LUFS, or Loudness Units Full Scale (also known as LKFS), is a standard unit of loudness measurement that accounts for human perception of sound. It provides a more accurate representation of perceived loudness than traditional peak meters, making it crucial for loudness normalisation in broadcast, streaming, and music production to achieve consistent audio levels.

LXF, or Leitch Exchange Format, is a proprietary file format developed by Leitch (now Imagine Communications) for professional video applications. It is used for storing and exchanging video and audio content within Leitch-based systems, often in broadcast environments.

M2TS is a container file format used for multiplexing audio, video and other streams. It is based on the MPEG Transport Stream container format and is commonly used for high-definition video on Blu-ray Discs and AVCHD.

An M3U8 file is a plain-text playlist file used in HLS (HTTP Live Streaming) that lists the media segments a player needs to reconstruct a video stream. It contains URIs to media segments and can also include information about different quality levels for adaptive bitrate streaming.

An MPD (Media Presentation Description) file is an XML-based manifest file used in MPEG-DASH that describes the structure and availability of media content. It provides information about the different bitrates, resolutions, and languages available, allowing the client to adapt the stream based on network conditions.

The central technical hub of a broadcast operation where all production signals converge, decisions are made, and the final output is controlled before transmission. The MCR is responsible for ensuring seamless playout, monitoring signal quality, and managing the overall broadcast workflow.

In HLS, a media playlist is a sub-playlist referenced by a master playlist. It describes a single rendition or media stream, containing a sequence of URIs to media segments (e.g., .ts files) that make up a portion of the video or audio content. The player downloads these segments sequentially to play the stream.

A media processing pipeline is a structured sequence of automated steps and components that transform raw or encoded media into a final, deliverable format for streaming. This pipeline typically includes encoding, transcoding, packaging, and DRM encryption, ensuring the content is optimized for various platforms and viewing conditions. It manages the entire lifecycle of media preparation from ingest to distribution.

In adaptive bitrate streaming protocols like HLS and MPEG-DASH, media segments are small, short pieces of a larger media stream or file. These segments are typically a few seconds long and are encoded at various bitrates, allowing the client player to dynamically switch between different quality levels based on network conditions to ensure smooth playback.

MediaInfo is a free, open-source program that provides technical and tag information about video and audio files. It analyzes multimedia files to display details such as codec, bitrate, resolution, frame rate, duration, and other technical parameters, making it an essential tool for professionals and enthusiasts to inspect and troubleshoot media files.

MediaSource Extensions (MSE) is a W3C API that extends the HTML5 `<video>` and `<audio>` elements, allowing JavaScript to dynamically construct media streams for playback. It enables web applications to feed byte streams to media codecs, facilitating advanced streaming techniques like adaptive bitrate streaming (e.g., HLS and DASH) and custom media processing directly within the browser. MSE is a cornerstone of modern web-based video players.

Mid-tier caching refers to the implementation of caching servers at an intermediate layer within a CDN's architecture, typically between the edge nodes and the origin server. This tiered approach helps to reduce the load on the origin server and improve cache hit ratios at the edge by serving as a larger, more centralized cache for less frequently accessed content.

MKV, or Matroska Video File, is an open-standard, free container format that can hold an unlimited number of video, audio, picture, or subtitle tracks in a single file. It is widely used for high-definition video content due to its flexibility and feature set.

Motion compensation is a technique used in inter-frame video compression that utilizes motion vectors obtained from motion estimation to predict the content of a current frame from a reference frame. By applying these predictions, only the residual differences need to be encoded, greatly improving compression efficiency.

Motion estimation is the process in video compression where the encoder identifies and quantifies the movement of objects or regions between successive video frames. It calculates motion vectors that describe the displacement of blocks of pixels, which are then used by motion compensation to predict frame content.

MOV is a multimedia container file format developed by Apple Inc. and is native to its QuickTime framework. It is commonly used for saving movies and other video files, supporting various tracks for video, audio, text, and effects.

MP4, or MPEG-4 Part 14, is a digital multimedia container format most commonly used to store video and audio, but can also be used to store other data such as subtitles and still images. It is based on the QuickTime File Format but adds support for MPEG features.

MPC-HBR, or Model Predictive Control - Hybrid Bitrate, refers to advanced ABR algorithms that utilize Model Predictive Control (MPC) techniques to make bitrate decisions. These algorithms predict future network conditions and buffer states to proactively select optimal bitrates, often combining throughput and buffer information. MPC-HBR aims to optimize Quality of Experience (QoE) by anticipating changes and making more informed streaming choices.

MPEG-1 Layer 3, commonly known as MP3, is a widely adopted lossy audio compression format. It significantly reduces file size by removing inaudible frequencies through perceptual coding, making it highly efficient for storing and transmitting digital audio, though at a compromise in audio fidelity compared to lossless formats.

MPEG-2 is a standard for the generic coding of moving pictures and associated audio information, widely used for digital television broadcasting (terrestrial, cable, and satellite) and DVD-Video. It combines lossy video and audio compression methods, supporting interlaced video formats common in analog broadcast systems.

MPEG-4 is a suite of international standards for compressing audio and visual digital data, defining formats for various multimedia applications. It encompasses multiple parts, including MPEG-4 Part 2 (Visual) for video compression and MPEG-4 Part 10 (AVC/H.264) for advanced video coding.

MPEG-DASH is an international standard for adaptive bitrate streaming that enables high-quality media content delivery over the internet using conventional HTTP web servers. It works by segmenting content into small, short segments available at various bit rates, allowing clients to adapt to changing network conditions.

MPEG Transport Stream (MPEG-TS) over UDP refers to the practice of transmitting MPEG-TS packets using the User Datagram Protocol. This method is widely used in broadcast and IPTV for its efficiency in delivering real-time audio and video, as UDP provides low overhead and speed, though without built-in reliability mechanisms.

Microsoft Smooth Streaming is an HTTP-based adaptive bitrate streaming protocol developed by Microsoft. It dynamically adjusts the video quality to match available bandwidth and CPU conditions, providing a seamless viewing experience. It uses fragmented MP4 (fMP4) for media delivery.

MSSSIM is an advanced perceptual video quality metric that extends the SSIM concept by evaluating structural similarity across multiple scales or resolutions of an image or video. This multi-scale approach provides a more robust and accurate assessment of perceived quality, as human visual perception operates at various scales, making it particularly useful for evaluating videos with different resolutions or viewing distances.

Multi-CDN is a strategy that involves using two or more Content Delivery Network providers simultaneously to deliver web content. This approach enhances performance, increases redundancy, and improves fault tolerance by distributing traffic across multiple networks, ensuring continuous content availability even if one CDN experiences an outage or performance degradation.

Multi-channel audio refers to audio systems that utilize more than two discrete audio channels to create a more immersive and spatial sound experience than traditional stereo. These systems typically involve multiple speakers strategically placed around the listener, with common configurations like 5.1 and 7.1 surround sound, to deliver distinct sound elements from various directions.

Multi-DRM refers to the strategy of implementing and managing multiple Digital Rights Management (DRM) systems simultaneously to protect content across a diverse ecosystem of devices and platforms. This approach is essential for content providers to reach the widest possible audience, as different devices and browsers often support specific DRM technologies (e.g., Widevine for Android/Chrome, FairPlay for Apple, PlayReady for Microsoft).

Multi-Period DASH is a feature within the MPEG-DASH standard that allows a video to be divided into multiple distinct Periods. Each Period can have its own set of properties, such as different content, advertisements, or even different encoding settings. This is particularly useful for inserting ads or structuring complex video presentations within a single manifest.

In HLS, a multi-variant playlist, also known as a master playlist, is the top-level index file that lists all available renditions (variant streams) of the content. It contains references to different media playlists, each representing a specific bitrate, resolution, or codec, allowing the player to select the most appropriate stream based on network conditions and device capabilities.

MXF, or Material Exchange Format, is a professional digital video and audio file format defined by SMPTE. It is designed to streamline the exchange of material between different professional video equipment, software, and facilities, supporting a wide range of codecs and metadata.

The server-side technique used by SSAI systems to modify HLS or DASH manifests in real time, replacing content segment references with ad segment references during ad breaks. The player receives a single, seamless manifest and is unaware that ad stitching has occurred, making the ad stream indistinguishable from content to ad blockers.

An advertisement inserted at a point within the body of a video, as opposed to before (pre-roll) or after (post-roll) the content. Mid-roll ads are triggered by SCTE 35 cue points in live streams or by timeline markers in VOD assets and are typically the highest-value ad placement due to viewer engagement levels.

MPEG-H Audio is an ISO/IEC standard for next-generation audio coding that supports object-based and scene-based audio, enabling personalised and immersive listening experiences. It is one of the mandatory audio codecs in ATSC 3.0 / NextGen TV and is also used in UHD Blu-ray and some OTT platforms. MPEG-H Audio allows broadcasters to deliver a single audio stream from which the receiver can render personalised mixes — adjusting dialogue levels, selecting language tracks, or enabling audio description — without requiring separate streams for each variant.

Network Device Interface (NDI) is a software specification developed by NewTek that enables high-definition video to be transmitted, received, and communicated between devices over a standard IP network. It allows multiple video-compatible devices to share large amounts of data with minimal lag, simplifying video production workflows.

An optional table within the MPEG Program Specific Information (PSI) or DVB Service Information (SI) that conveys information about the physical organization of the broadcast network. The NIT provides details about transport streams, frequencies, and network characteristics, helping receivers to tune into and navigate available services.

NVENC (NVIDIA Encoder) is a dedicated hardware encoder integrated into NVIDIA graphics cards. It offloads the computationally intensive task of video encoding from the CPU, allowing for faster encoding speeds and reduced system resource usage, particularly beneficial for live streaming and video recording without significantly impacting gaming or other CPU-intensive applications.

The consumer-facing brand name for ATSC 3.0, the next-generation over-the-air broadcast television standard in North America. NextGen TV-certified devices can receive ATSC 3.0 signals and support features such as 4K UHD, HDR, immersive audio, personalised content, and emergency alerts with rich media. The ATSC 3.0 / NextGen TV rollout is ongoing across major US markets, with broadcasters simulcasting on both ATSC 1.0 and ATSC 3.0 during the transition period.

Object-based audio is an advanced audio mixing and rendering paradigm where individual sound elements (objects) are treated as discrete entities with associated metadata, rather than being pre-mixed into fixed channels. This allows for dynamic rendering of sounds in a three-dimensional space, adapting to various playback systems and creating highly personalized and immersive listening experiences.

Ogg is a free, open standard container format maintained by the Xiph.Org Foundation. It is designed to provide efficient streaming and manipulation of high-quality digital multimedia, commonly used with Vorbis audio and Theora video codecs.

An Open GOP is a Group of Pictures where frames within the GOP may reference frames from previous GOPs for decoding. While potentially offering slightly better compression due to broader reference possibilities, open GOPs can complicate random access and editing, as decoding a frame might require data from an earlier GOP.

Opus is a versatile, open-source, and royalty-free audio codec designed for interactive speech and music transmission over the internet. It combines the best features of speech-oriented codecs and high-fidelity music codecs, offering excellent quality across a wide range of bitrates and applications, from voice calls to live streaming.

The origin server is the primary server where the original, authoritative version of content resides. In a CDN setup, when an edge node does not have the requested content in its cache (a cache miss), it retrieves the content from the origin server. All content updates and modifications are initially made on the origin server.

An origin server is the primary server where original content, such as video files and web assets, is stored and managed. In streaming workflows, it receives encoded video segments and playlist files, processing and responding to incoming internet requests from clients or content delivery networks (CDNs). It serves as the authoritative source for content before distribution.

OTT, or Over-the-Top, refers to the delivery of video and audio content directly to viewers over the internet, bypassing traditional broadcast, cable, or satellite television providers. Services like Netflix and Hulu are examples of OTT platforms, offering content directly to internet-connected devices. This model provides consumers with greater flexibility and choice in their media consumption.

An IAB standard that provides a common API for third-party ad measurement and verification SDKs to access data about video ad playback environments — including viewability, audio state, and player state — without requiring custom integrations per platform. OMID is widely used in CTV and mobile video advertising.

An IAB standard protocol for real-time bidding (RTB) between DSPs and SSPs. OpenRTB defines the JSON-based bid request and bid response format used in programmatic advertising auctions, including extensions for CTV and OTT inventory such as content metadata, device type, and SSAI compatibility signals.

OFDM is a digital modulation technique that transmits data by splitting a high-speed signal across many closely spaced, orthogonal sub-carrier frequencies simultaneously. It is highly resistant to multipath interference and frequency-selective fading, making it the modulation method of choice for modern terrestrial broadcast standards including DVB-T2 and ATSC 3.0. OFDM is also the foundation of Wi-Fi (802.11) and LTE/5G mobile networks.

A P-frame, or Predicted frame, is a video frame that stores only the changes from a preceding I-frame or another P-frame. It achieves better compression than an I-frame by using motion estimation and compensation to predict its content from a previous reference frame.

P2 MXF refers to the Material Exchange Format (MXF) files recorded on Panasonic P2 cards. P2 is a professional solid-state memory card format used in Panasonic camcorders, and the MXF files contain video, audio, and metadata in a broadcast-friendly wrapper.

P3, often referred to as DCI-P3 or Display P3, is a wide-gamut RGB color space primarily used in digital cinema projection and increasingly adopted in consumer electronics for HDR content. It offers a significantly larger range of colors than the sRGB/BT.709 standard, particularly in reds and greens, contributing to a more vibrant and lifelike visual experience.

A packager is a component in a streaming workflow that takes encoded video streams and prepares them for delivery to various devices and platforms. This process typically involves segmenting the video into small chunks and generating manifest files (e.g., HLS or MPEG-DASH) that describe the available bitrates and segment order. Packaging ensures compatibility and adaptive streaming capabilities for different player technologies.

A fundamental table within the MPEG Program Specific Information (PSI) that lists all programs available in a transport stream and their corresponding Program Map Table (PMT) Packet Identifiers (PIDs). The PAT is essential for a receiver to locate the PMT for any given program, enabling the decoding of its elementary streams.

PCM, or Pulse Code Modulation, is a method used to digitally represent analog signals. It involves sampling the analog signal at regular intervals and quantizing each sample into a binary value. PCM is the standard form of digital audio in computers, compact discs, and other digital audio systems, forming the basis for many other audio formats.

Per-scene encoding, also known as Content-Aware Encoding, dynamically adjusts encoder settings on a scene-by-scene or shot-by-shot basis within a single video file. By analyzing the spatial and temporal complexity of each scene, the encoder allocates more bits to complex, high-motion sequences and fewer bits to static scenes, maximizing overall visual quality while minimizing file size.

Per-title encoding is an advanced video compression technique that analyzes the complexity of each individual video asset to generate a customized encoding ladder. Instead of using a static, one-size-fits-all set of bitrates and resolutions, it assigns optimal encoding parameters tailored to the specific content, saving bandwidth on simple videos like animations while preserving quality for complex, high-motion content.

Perceptual coding is a lossy compression technique that exploits the limitations and characteristics of the human visual system to discard information that is least likely to be perceived by viewers. By prioritizing the retention of visually important details and removing redundant or imperceptible data, it achieves significant compression ratios while maintaining a subjectively high quality of experience.

A 13-bit code in the header of an MPEG transport stream packet that uniquely identifies the elementary stream or data table to which the packet belongs. PIDs are fundamental for demultiplexing, allowing a receiver to distinguish between different video, audio, and data components within a single transport stream.

A player SDK (Software Development Kit) is a collection of software tools, libraries, and APIs that developers use to integrate video playback functionality into their applications or websites. These SDKs provide pre-built components for handling video decoding, rendering, adaptive streaming, and user interface controls, simplifying the development of custom video players. They abstract away the complexities of media playback.

Software systems and processes that manage and automate the broadcasting of video and audio content for television or radio channels. It ensures that all media elements, including programs, commercials, and graphics, are delivered in the correct order and at the scheduled time, significantly streamlining broadcast operations.

PlayReady is a Digital Rights Management (DRM) technology developed by Microsoft, designed to protect audio and video content across a wide range of devices, including Windows devices, Xbox consoles, and many Smart TVs. It secures content during distribution and playback through encryption and license management, ensuring compliance with content owners' rights.

A table within the MPEG Program Specific Information (PSI) that provides detailed information about a specific program, including the Packet Identifiers (PIDs) of its elementary video, audio, and data streams. The PMT also contains descriptors for the program and its elementary streams, allowing the receiver to properly decode and present the content.

A Point of Presence (PoP) is a data center or facility within a CDN's distributed network where edge servers are located. PoPs serve as access points for users, enabling content to be delivered from a location geographically closer to them, which reduces latency and enhances the user experience. Each PoP typically contains multiple servers and networking equipment.

Perceptual Quantizer (PQ), standardized as SMPTE ST 2084, is an HDR transfer function that maps image signal values to absolute light levels, enabling displays to reproduce a wide range of brightness up to 10,000 nits. Unlike traditional gamma curves used in SDR, PQ is designed to be perceptually uniform, meaning equal steps in the PQ code value correspond to equal perceptual differences in brightness, making it ideal for HDR content mastering.

Apple ProRes is a family of lossy, perceptually lossless intermediate video codecs developed by Apple for use in post-production workflows. It employs intra-frame compression, allowing for excellent random access performance during editing and maintaining high quality while reducing storage requirements compared to uncompressed video.

Metadata within an MPEG transport stream that describes the structure and content of programs (channels), enabling a receiver to demultiplex and decode the various elementary streams. PSI includes four key tables: PAT, CAT, PMT, and NIT, which provide information about program associations, conditional access, program mapping, and network details.

PSNR is a traditional engineering metric used to measure the objective quality of a compressed video by calculating the ratio between the maximum possible power of a signal and the power of corrupting compression noise. While widely used due to its mathematical simplicity, PSNR measures absolute pixel differences rather than perceived visual quality, meaning it does not always align with how human eyes perceive video degradation.

A PSSH (Protection System Specific Header) box is a standardized container within an ISO Base Media File Format (ISOBMFF) that holds metadata specific to a particular content protection system (DRM). It provides information that a client device needs to interact with a DRM license server, such as the System ID of the DRM and any proprietary data required for license acquisition and content decryption. PSSH boxes are crucial for multi-DRM interoperability.

PVOD, or Premium Video On Demand, is a specific type of TVOD where newly released films or exclusive content are made available for rental or purchase at a premium price, often coinciding with or preceding their traditional theatrical release or standard VOD window. This model allows consumers early access to high-demand content from the comfort of their homes. It is characterized by its higher price point and early availability.

An advertisement played after the main video content has ended. Post-roll ads have the lowest completion rates of the three standard placement types (pre-roll, mid-roll, post-roll) because many viewers abandon playback before the content ends.

An advertisement played before the main video content begins. Pre-roll ads typically have the highest completion rates of any ad placement type and are the most common format in OTT and streaming environments. They are delivered via CSAI or SSAI and defined in VAST responses.

The automated buying and selling of digital advertising inventory using software and real-time bidding, as opposed to direct insertion orders. In streaming, programmatic advertising connects OTT publishers (via SSPs) with advertiser demand (via DSPs) through OpenRTB auctions, enabling dynamic, targeted ad delivery at scale.

PSIP is an ATSC standard that defines how programme guide and system information is delivered within an ATSC 1.0 digital television broadcast. It provides the data that populates the on-screen electronic programme guide (EPG) on ATSC receivers, including channel names, programme titles, descriptions, ratings, and scheduling information. PSIP is the ATSC equivalent of DVB's SI (Service Information) tables.

QoE, or Quality of Experience, is a subjective measure of a user's overall satisfaction with a streaming service, encompassing both technical performance and perceived quality. It considers factors like video startup time, rebuffering frequency, video quality, and overall responsiveness, reflecting how well the service meets viewer expectations. QoE is a holistic view of the user's interaction with the streaming content.

QoS, or Quality of Service, refers to the technical metrics that measure the performance of the underlying network and infrastructure delivering a streaming service. These metrics include bandwidth, latency, jitter, and packet loss, which directly impact the technical delivery of content. QoS focuses on the network's ability to provide a consistent and reliable stream.

Quantisation is a lossy compression technique in video encoding that reduces the precision of transformed coefficients (e.g., from DCT) by mapping a range of input values to a single output value. This process discards less perceptually significant information, thereby reducing the data size, with higher quantisation levels leading to greater compression but also more loss of detail.

QUIC (Quick UDP Internet Connections) is a general-purpose transport layer network protocol initially designed by Google, built on top of UDP. It aims to reduce latency and improve performance for web applications by providing multiplexed connections, stream-level flow control, and built-in encryption, serving as the foundation for HTTP/3.

QUIC/HTTP3 CDN refers to Content Delivery Networks that support and leverage the QUIC (Quick UDP Internet Connections) protocol, which forms the basis of HTTP/3. QUIC aims to improve web performance by reducing connection establishment overhead, eliminating head-of-line blocking, and providing better loss recovery, resulting in faster and more reliable content delivery, especially over unreliable networks.

Intel Quick Sync Video (QuickSync) is Intel's hardware video encoding and decoding technology integrated into some of its CPUs and integrated GPUs. It accelerates video processing tasks like transcoding, allowing for faster performance and lower CPU utilization compared to software-based encoding, making it suitable for applications such as media playback, streaming, and video editing.

Rate control in video compression refers to the algorithms and strategies used to manage the bitrate of an encoded video stream to meet specific targets for file size, bandwidth, and quality. Different methods, such as Constant Bit Rate (CBR), Variable Bit Rate (VBR), Constant Rate Factor (CRF), Constant Quantization Parameter (CQP), and Adaptive Bit Rate (ABR), offer varying trade-offs between these factors.

RDD9, or Registered Disclosure Document 9, is a SMPTE specification that defines the MXF Interoperability Specification for Sony MPEG Long GOP Products. It outlines how MPEG-2 Picture (ES), AES3 audio, and ANC packets are mapped into the MXF Generic Container for these specific products.

Rebuffering ratio is a key Quality of Experience (QoE) metric that measures the percentage of total playback time during which the video stream is paused due to insufficient buffered data. A high rebuffering ratio indicates frequent interruptions and a poor viewing experience, often caused by network congestion or inadequate bandwidth. Minimizing this ratio is crucial for viewer retention.

Reference frames are previously encoded and reconstructed video frames that are used by P-frames and B-frames to predict their content during inter-frame compression. By referring to these frames, encoders can store only the differences, leading to significant data reduction.

Video resolution refers to the number of distinct pixels that compose each frame of a video, typically expressed as width × height (e.g., 1920x1080 for Full HD). A higher resolution indicates a greater number of pixels, resulting in a sharper, more detailed image, while lower resolutions offer less detail but require less bandwidth and storage.

Reliable Internet Stream Transport (RIST) is an open-source, open-specification transport protocol designed for reliable transmission of video over lossy IP networks. It uses a NACK-based Selective Retransmission protocol to recover from packet loss, ensuring broadcast-grade video quality over the internet.

RTP Control Protocol (RTCP) works in conjunction with RTP to provide out-of-band control information and statistical feedback on the quality of media distribution during a session. It helps monitor call quality in VoIP and video by sending data like packet loss and round-trip time back to the sender.

Real-Time Messaging Protocol (RTMP) is a communication protocol for streaming audio, video, and data over the Internet, originally developed by Macromedia (later Adobe). It is effective in keeping low latency with stable connections, making it suitable for live video streaming applications.

Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks, focusing on the streaming and timing of media data. It provides end-to-end network transport functions suitable for applications transmitting real-time data, often working in conjunction with RTCP.

Real-Time Streaming Protocol (RTSP) is an application-level network protocol designed for controlling the delivery of data with real-time properties, primarily used for establishing and controlling media sessions between endpoints. It acts as a "network remote control" for streaming media servers, separating signaling from media transport.

A detailed, item-by-item sequence of events or a script that outlines the content and timing of a broadcast program, such as a news show or live event. It serves as a blueprint for the production team, ensuring that all segments, cues, and transitions occur in the correct order and at the scheduled times.

An auction mechanism in programmatic advertising in which ad impressions are bought and sold in real time — typically within 100 milliseconds of a viewer initiating a stream. Each available impression is auctioned to the highest bidder among competing DSPs, with the winning creative being stitched into the stream by the SSAI system.

SAMPLE-AES is an encryption method used in HTTP Live Streaming (HLS) where individual media samples (e.g., video frames or audio packets) are encrypted using the Advanced Encryption Standard (AES). Unlike AES-128 HLS encryption, which encrypts entire media segments, SAMPLE-AES offers more granular control over encryption, allowing for specific portions of the media to be protected. This method is often associated with Apple's FairPlay Streaming DRM.

Serial Digital Interface (SDI) is a family of digital video interfaces standardized by SMPTE, primarily used for transmitting uncompressed digital video signals over coaxial or fiber optic cables within broadcast facilities. It provides a robust and reliable method for connecting various pieces of professional video equipment.

Standard Dynamic Range (SDR) refers to the conventional video technology that has been used for decades, characterized by a limited range of luminance and color compared to HDR. SDR content is typically displayed on standard televisions and monitors, offering a narrower contrast ratio and color gamut, which results in a less immersive visual experience than HDR.

A table within DVB Service Information (SI) that provides descriptive data about the services available in a transport stream, including service names, service providers, and service types (e.g., television, radio). The SDT helps receivers present a clear and comprehensive list of available channels to the user.

Shaka Player is an open-source JavaScript library developed by Google for adaptive media streaming, primarily supporting DASH and HLS formats in web browsers without requiring plugins. It leverages MediaSource Extensions (MSE) and Encrypted Media Extensions (EME) to handle playback of adaptive and DRM-protected content. Shaka Player is known for its robust feature set and wide compatibility.

Metadata embedded within a digital broadcast stream (e.g., DVB systems) that provides essential information about the services available to the receiver. SI includes details such as program schedules, service names, and event information, enabling receivers to present Electronic Programme Guides (EPGs) and properly decode services.

Signed URLs are specially constructed URLs that provide temporary, authenticated access to private content stored on a CDN. These URLs include cryptographic signatures, expiration times, and sometimes IP restrictions, allowing content providers to grant limited-time access to specific users without making the content publicly available.

Simulcrypt is a Digital Video Broadcasting (DVB) protocol that enables multiple Conditional Access Systems (CAS) to encrypt and control access to the same broadcast content simultaneously. This allows broadcasters to serve different subscriber bases, each using a distinct CAS, from a single content stream, optimizing infrastructure and content delivery for diverse markets or service tiers.

In the context of conditional access systems (CAS) for broadcast television, a smartcard is a physical card containing an embedded microchip that securely stores subscriber information, decryption keys, and entitlements. It is inserted into a set-top box or integrated digital television (IDTV) to authenticate the user and enable the decryption of encrypted broadcast content, ensuring that only paying subscribers can access premium channels and services.

SMPTE ST 2022-6 is a standard that specifies the encapsulation of a complete SDI signal (including video, audio, and ancillary data) into a single IP stream for transport over IP networks. It allows for the transmission of uncompressed high-definition video streams over standard network infrastructure.

SMPTE ST 2022-7 defines a seamless protection switching mechanism for IP-based media transport, enabling redundant transmission of identical media data streams over two separate network paths. This ensures uninterrupted delivery and robust error recovery in case of network failures, crucial for broadcast reliability.

SMPTE ST 2110-20 specifies the transport of uncompressed active video over IP networks using the Real-time Transport Protocol (RTP). It allows for the independent routing and processing of video essence, supporting various video formats and resolutions for high-quality broadcast production.

SMPTE ST 2110-30 specifies the transport of PCM digital audio streams over IP networks, referencing AES67 for interoperability. It enables the independent routing and processing of audio essence, providing flexibility for audio mixing and processing in broadcast workflows.

SMPTE ST 2110-40 specifies the transport of ancillary data, such as timecode, closed captions, and other metadata, over IP networks. It allows ancillary data to be transported independently of video and audio, providing flexibility for workflows like live closed captioning and subtitling.

A set of cooperating standards defined by the Society of Motion Picture and Television Engineers (SMPTE) to label individual frames of video or film with a timecode. It provides a precise time reference for editing, synchronization, and identification of media content, making modern videotape editing and non-linear editing systems possible.

Software encoding uses a computer's central processing unit (CPU) to perform video compression. While generally more flexible and capable of achieving higher quality at lower bitrates than hardware encoding, it is also more computationally intensive, consuming significant CPU resources and often resulting in slower encoding speeds. It is preferred for tasks where quality and flexibility are paramount, such as professional video production.

Secure Reliable Transport (SRT) is an open-source video transport protocol that utilizes UDP to provide reliability and security optimized for low-latency live video streaming across unpredictable networks. It securely and reliably streams video content with AES encryption and packet loss recovery.

SRT Caller/Listener mode describes the connection establishment mechanism in Secure Reliable Transport (SRT), where one endpoint acts as the 'Caller' to initiate the connection, and the other acts as the 'Listener' to await incoming connections. This mode is crucial for traversing firewalls and establishing reliable links between SRT sources and destinations.

SRT Rendezvous is a connection mode in Secure Reliable Transport (SRT) where both the sender and receiver initiate an outgoing connection to each other simultaneously. This mode is particularly useful for establishing connections when both endpoints are behind firewalls or NATs, without requiring port forwarding.

SSIM is a perceptual video quality metric designed to measure the structural similarity between an encoded video and its uncompressed reference source. Unlike PSNR, which measures absolute pixel errors, SSIM models human visual perception by evaluating changes in luminance, contrast, and structural information, making it a more accurate predictor of subjective video quality.

SMPTE ST 2022 is a suite of standards that describes how to send digital video and ancillary data over IP networks, providing a reliable way to transmit signals traditionally sent over serial interfaces. It includes various parts for different types of media and error correction mechanisms.

SMPTE ST 2110 is a suite of standards that defines the real-time transport of professional media over IP networks, separating video, audio, and ancillary data into independent, synchronized streams. This essence-based approach provides greater flexibility and efficiency for broadcast production and distribution facilities.

Stall events, often referred to as buffering events, occur when the video playback pauses unexpectedly because the player's buffer has emptied and is waiting for more data to download. These interruptions significantly degrade the Quality of Experience (QoE) for viewers. Frequent stall events are typically indicative of network issues, insufficient bandwidth, or inefficient content delivery.

Startup time, also known as video startup time (VST) or join time, is the duration from when a user initiates playback (e.g., by pressing play) to when the first frame of the video appears on the screen. It is a critical Quality of Experience (QoE) metric, as longer startup times can lead to viewer abandonment. Optimizing startup time involves efficient content delivery and player initialization.

Stream health monitoring involves continuously tracking and analyzing various metrics related to the performance and quality of a video stream, from ingest to playback. This includes monitoring for issues like buffering, high startup times, low bitrates, and error rates to ensure a smooth and high-quality viewing experience. Proactive monitoring helps identify and resolve problems before they impact a large audience.

Super-resolution AI upscaling utilizes artificial intelligence and machine learning algorithms to increase the resolution of video footage beyond simple pixel interpolation. Unlike traditional upscaling, AI upscaling can intelligently generate new pixel data, reconstruct fine details, and enhance overall visual quality, making lower-resolution content appear significantly sharper and more detailed on high-resolution displays.

SVOD, or Subscription Video On Demand, is a monetization model where users pay a recurring subscription fee to gain unlimited access to a library of video content. This model provides an ad-free viewing experience and allows subscribers to watch content at their convenience. Popular examples include Netflix and Disney+.

SVT-AV1 (Scalable Video Technology for AV1) is an open-source, software-based AV1 video encoder developed by Intel and Netflix. It is designed for high-performance and scalable AV1 encoding, offering excellent compression efficiency and visual quality, particularly for demanding applications like video-on-demand and live streaming. SVT-AV1 is known for its ability to scale across multiple CPU cores, making it a powerful solution for software-based AV1 encoding.

A digital program insertion standard published by the Society of Cable Telecommunications Engineers (SCTE) that defines the binary cue message format used to signal ad break opportunities in MPEG-2 transport streams. A SCTE 35 splice_insert() command carries timing, duration, and segmentation descriptor information that downstream SSAI systems use to trigger ad decisioning and manifest manipulation.

An SCTE standard that defines the interface between automation systems (such as master control or playout servers) and SCTE 35 cue tone inserters. SCTE 104 messages are sent over IP from the automation system to the encoder, which converts them into SCTE 35 binary cue messages embedded in the transport stream.

An SCTE standard for event scheduling and notification interface (ESNI) that defines an XML-based protocol for communicating programme schedule and ad break information between content owners, distributors, and ad systems over IP networks. SCTE 224 complements SCTE 35 by providing richer metadata about upcoming ad opportunities.

An IAB standard that replaces VPAID, providing a secure, sandboxed API for interactive ad experiences in video players. SIMID allows ad creatives to communicate with the player to access playback state, trigger interactions, and display overlays, without executing arbitrary JavaScript in the player context.

An ad insertion architecture in which advertisements are stitched into the video stream on the server before delivery to the client. The player receives a single, seamless stream containing both content and ads, making the ad segments indistinguishable from content and effectively bypassing client-side ad blockers. SSAI systems handle manifest manipulation, ad transcoding, bitrate matching, and beacon firing.

A programmatic advertising platform used by publishers and content owners to manage, sell, and optimise their ad inventory across multiple demand sources (DSPs and ad networks). In OTT and CTV, SSPs aggregate demand and run real-time auctions to maximise yield on available ad impressions.

A broadcast network architecture in which all transmitters in a coverage area operate on the same radio frequency and are precisely synchronised to transmit identical signals simultaneously. SFNs are used in DVB-T2 and ATSC 3.0 deployments to maximise spectrum efficiency and coverage, as the signals from multiple transmitters combine constructively at the receiver rather than causing interference.

Telecine is the process of converting motion picture film into a video format for television broadcast or digital distribution. This involves adjusting the frame rate of film (typically 24 frames per second) to match the video frame rate (e.g., 29.97 fps for NTSC or 25 fps for PAL), often using techniques like 3:2 pulldown, which can introduce artifacts.

Theora is a free and open video compression format developed by the Xiph.Org Foundation, based on On2 Technologies' VP3 codec. It is typically used within the Ogg container format, offering royalty-free video for web and multimedia applications, and is often paired with Vorbis audio.

Throughput-based ABR (Adaptive Bitrate) algorithms make decisions about which video rendition to stream primarily by estimating the available network bandwidth or throughput. These algorithms continuously measure the download speed of video segments and select a bitrate that is slightly below the estimated throughput to avoid buffering. While effective, they can sometimes be reactive to sudden network fluctuations.

TIFO, or Telestream Intermediate Format, is a proprietary intermediate file format wrapper developed by Telestream. It is primarily used for integration between Telestream Pipeline and Telestream transcoding products, preserving VBI/VANC data within a Data/Control track.

Tone mapping is a technique used in video processing to convert High Dynamic Range (HDR) content to Standard Dynamic Range (SDR) displays, or to adapt a wide range of brightness and color values to a more limited display capability. It intelligently compresses the dynamic range while preserving visual details and perceived contrast, ensuring that HDR content can be viewed on SDR screens without appearing washed out or overly dark.

A transcoder is a specialized encoder that converts an already encoded video or audio file from one format or bitrate to another. This process is essential for adaptive bitrate streaming, allowing content to be delivered in multiple quality levels to suit various network conditions and device capabilities. Transcoding ensures optimal viewing experiences across diverse environments.

Transcoding is the process of taking an already encoded digital media file, decoding it, and re-encoding it into a different format, codec, bitrate, or resolution. This computationally intensive process is essential for ensuring video content is compatible with a wide variety of playback devices and network conditions.

Transizing is a specific type of transcoding that alters the spatial resolution or frame size of a video file, such as downscaling a 4K UHD video to 1080p or 720p. This process is typically performed alongside transrating to create the various quality levels required for an adaptive bitrate streaming ladder.

Transmuxing, or transcode-multiplexing, is the process of changing the container format of a digital media file without altering the underlying audio or video codecs. It repackages the existing encoded data into a new delivery format, such as converting an MP4 file to an HLS stream, which requires significantly less computational power than full transcoding.

Transrating is a specific type of transcoding that involves changing the bitrate of a video file while maintaining its original resolution and codec. It is commonly used to create lower-bitrate renditions of a high-quality source file, enabling adaptive bitrate streaming for viewers with limited bandwidth.

True Peak refers to the actual peak level of an audio signal, including inter-sample peaks that may occur between digital samples. Unlike traditional sample peak meters, True Peak meters can detect these hidden peaks, which are crucial for preventing clipping and distortion when the digital signal is converted back to analog or processed further.

TS, or MPEG Transport Stream, is a standard digital container format for transmission and storage of audio, video, and Program Specific Information (PSI) data. It is commonly used in broadcast systems like DVB and ATSC, and for Blu-ray Disc video.

Time to Live (TTL) is a setting that determines how long a piece of content is stored in a CDN's cache before it is considered stale and must be re-validated or re-fetched from the origin server. A shorter TTL ensures content freshness but can increase origin server load, while a longer TTL reduces origin requests but risks serving outdated content.

TVOD, or Transactional Video On Demand, is a monetization model where users pay a one-time fee to access a specific piece of video content, either for rental (temporary access) or purchase (permanent ownership). This model is commonly used for new movie releases or premium content that is not part of a subscription library. It offers flexibility for consumers to pay only for the content they wish to watch.

Two-pass encoding is a video encoding technique that involves processing the video file twice to optimize compression efficiency and quality. In the first pass, the encoder analyzes the video content to gather statistics about its complexity, and in the second pass, it uses this information to intelligently allocate bits, resulting in a more consistent quality and smaller file size for a given target bitrate.

The process of re-encoding advertisement creative assets to match the codec, resolution, bitrate, frame rate, and segment duration of the content stream into which they will be stitched. Mismatched ad assets cause buffering, visual artefacts, or decoder errors at splice points; SSAI platforms typically maintain a just-in-time transcoding pipeline to normalise ad assets on receipt.

Ultra-low latency streaming refers to video delivery with an extremely short delay, typically under one second (often 500 milliseconds or less), from the moment an event is captured by a camera to when it appears on a viewer's screen. This level of latency is crucial for interactive applications, live sports betting, gaming, and real-time communication, where immediate feedback and synchronization are paramount. Technologies like WebRTC and HESP are designed to achieve ultra-low latency.

Upscaling is the process of increasing the resolution of a video or image to a higher resolution, typically to match the native resolution of a display device. This technique interpolates missing pixel data to enlarge the image, aiming to improve visual clarity and detail when a lower-resolution source is viewed on a higher-resolution screen. While it can make content appear sharper, it does not add true detail that was not present in the original source.

Non-video information embedded within the vertical blanking interval of a digital video signal, typically in SDI (Serial Digital Interface) streams. VANC carries various types of metadata, such as closed captions, audio, timecode, and active format description, without interfering with the active video content.

Variable Bit Rate (VBR) is a rate control method where the encoder adjusts the bitrate dynamically based on the complexity of the video content. More bits are allocated to complex scenes to preserve quality, and fewer bits to simpler scenes, resulting in a more consistent visual quality and generally smaller file sizes compared to CBR for a given quality level.

VC-1 (SMPTE 421M) is a video coding format that evolved from Microsoft's Windows Media Video 9, officially approved as a SMPTE standard. It was marketed as a lower-complexity alternative to H.264/MPEG-4 AVC, supporting both interlaced and progressive video, and was adopted by HD DVD and Blu-ray Disc formats.

Video.js is a popular open-source HTML5 video player framework written in JavaScript, designed to provide a consistent and customizable video playback experience across various web browsers and devices. It supports adaptive streaming formats like HLS and MPEG-DASH through plugins and offers a rich API for developers to extend its functionality. Video.js is widely used for embedding and controlling video content on websites.

VideoToolbox is a low-level framework provided by Apple that offers direct access to hardware-accelerated video encoding and decoding capabilities on macOS and iOS platforms. It enables developers to perform video compression and decompression efficiently, leveraging dedicated hardware for improved performance and reduced power consumption in applications like video editing, streaming, and real-time communication.

Visible watermarking involves embedding a clearly discernible overlay, such as a logo, text, or pattern, directly onto video content. Unlike forensic watermarking, its purpose is to act as a deterrent against unauthorized copying and distribution by making the content less desirable for piracy. While it doesn't prevent copying, it publicly asserts ownership and can discourage illicit use.

A form of SMPTE timecode encoded on one scan line within the vertical blanking interval of a video signal. The primary advantage of VITC over LTC is its ability to be read even when the video tape is stationary or paused, making it useful for precise frame identification.

VMAF is an objective, full-reference video quality assessment algorithm developed by Netflix that combines human vision modeling with machine learning to predict subjective visual quality. It evaluates video quality by comparing the encoded video against the pristine source, taking into account scaling artifacts, compression artifacts, and temporal characteristics to produce a score that closely correlates with human perception.

A VOD (Video On Demand) workflow describes the sequence of operations involved in preparing and delivering pre-recorded video content for on-demand access by viewers. This typically includes ingest, encoding, transcoding into multiple renditions, packaging, storage, and distribution through a CDN. Unlike live streaming, VOD content is available for playback at any time chosen by the user.

Vorbis is a free, open-source, and patent-free lossy audio compression format developed by the Xiph.Org Foundation. It is typically used within the Ogg container format (Ogg Vorbis) and aims to provide competitive audio quality and compression efficiency without licensing restrictions, serving as an alternative to proprietary codecs like MP3.

VP8 is an open and royalty-free video compression format initially developed by On2 Technologies and later released by Google. It is primarily used for real-time applications such as WebRTC and as a replacement for GIF, offering efficient compression for progressive scan video.

VP9 is an open and royalty-free video compression standard developed by Google, serving as the successor to VP8. It offers improved compression efficiency compared to H.264/AVC and is widely used for online video platforms like YouTube and Netflix, supporting resolutions up to 8K.

An IAB standard XML schema that defines the structure of an ad server response to a video ad request. A VAST response contains the ad creative URI, duration, click-through URL, and a set of tracking event beacons (impression, quartile, complete, etc.). VAST 4.x introduced support for server-side ad verification and mezzanine file delivery.

A metric that measures whether an ad was actually seen by a viewer, as defined by the IAB/MRC standard: at least 50% of the ad's pixels visible for at least two consecutive seconds (display) or with audio on and at least 50% of pixels visible for at least two seconds (video). In OTT and CTV, viewability measurement is performed using OMID-compliant SDKs.

An IAB standard XML schema that defines the placement of multiple ad breaks within a single video asset, specifying the time offset, break type (pre-roll, mid-roll, post-roll), and VAST ad tag URL for each break. VMAP is used in VOD workflows to describe the full ad schedule for a piece of content in a single response.

A legacy IAB standard that allowed JavaScript-based interactive ad creatives to communicate with the video player, enabling rich interactive experiences and third-party measurement. VPAID has been deprecated in favour of SIMID due to security concerns (arbitrary JS execution in the player context) and incompatibility with SSAI environments.

WebM is an open, royalty-free media file format designed for the web. It is based on a subset of the Matroska container format and uses VP8 or VP9 video codecs and Vorbis or Opus audio codecs, offering high-quality video streaming with low computational overhead.

Web Real-Time Communication (WebRTC) is an open-source project that enables web browsers and mobile applications to communicate in real-time via simple APIs. It supports peer-to-peer communication for audio, video, and data, facilitating low-latency interactive applications without the need for plugins.

WebRTC latency refers to the minimal delay inherent in real-time communication established using the Web Real-Time Communication (WebRTC) protocol. Designed for peer-to-peer data exchange, WebRTC typically achieves very low latencies, often in the range of tens to hundreds of milliseconds, making it ideal for applications requiring instantaneous audio, video, and data transfer, such as video conferencing and interactive live streaming. It is a key technology for ultra-low latency scenarios.

WebRTC-HTTP Egress Protocol (WHEP) is an HTTP-based protocol designed to enable WebRTC-based viewers to consume content from streaming services or Content Delivery Networks. It acts as a companion to WHIP, standardizing the egress (playback) side of WebRTC streaming workflows.

WebRTC-HTTP Ingest Protocol (WHIP) is a simple HTTP-based protocol that allows WebRTC-based encoders or publishers to ingest live media content into streaming services or Content Delivery Networks. It standardizes the signaling process for WebRTC ingest, replacing custom ad-hoc protocols with a unified approach.

Wide colour gamut (WCG) refers to a display or video system's ability to reproduce a significantly larger range of colors than traditional Standard Dynamic Range (SDR) systems, typically encompassing color spaces like DCI-P3 and BT.2020. WCG is a key component of High Dynamic Range (HDR) content, enabling more vibrant, lifelike, and accurate color reproduction for a richer visual experience.

Widevine is a Digital Rights Management (DRM) technology developed by Google, widely used by major streaming services like Netflix, Amazon Prime Video, and YouTube. It secures premium content across various platforms and devices by encrypting media and managing decryption keys, ensuring that only authorized users can access the content.

A sequential ad decisioning strategy in which ad requests are sent to demand sources one at a time in priority order, moving to the next source only if the previous one returns no fill. Waterfall is being replaced by header bidding and unified auction approaches that solicit bids from all demand sources simultaneously to maximise yield.

x264 is a free and open-source software library and command-line utility that encodes video streams into the H.264/MPEG-4 AVC video compression format. It is widely recognized for its efficiency and high-quality output, making it a popular choice for various applications, including internet video streaming, Blu-ray disc encoding, and professional video production.

x265 is a free and open-source software library for encoding video streams into the High Efficiency Video Coding (HEVC), also known as H.265, compression format. It offers significantly improved compression efficiency compared to its predecessor, H.264, enabling smaller file sizes at equivalent visual quality or higher quality at the same bitrate, making it ideal for 4K/UHD content and bandwidth-constrained environments.

XDCAM is a series of professional video products introduced by Sony, utilizing optical disc, solid-state memory cards, or hard disk drives for recording. While not a container format itself, XDCAM products typically record video in MXF or MP4 containers with specific codecs like MPEG HD422.

Extended High-Efficiency Advanced Audio Coding (xHE-AAC) is the latest evolution in the AAC family, designed for adaptive streaming and digital radio across a wide range of bitrates. It provides excellent sound quality for both speech and music, with features like loudness and dynamic range control, making it robust for varying network conditions.

Zixi is a proprietary software-defined video platform (SDVP) and protocol designed for broadcast-quality live video delivery over IP networks. It provides reliability, security, and ultra-low latency by optimizing bandwidth and offering features like error-free streaming and adaptive bitrate.