HTTP Live Streaming
HTTP Live Streaming (also known as HLS) is an HTTP-based adaptive bitrate streaming communications protocol developed by Apple Inc. and released in 2009. Support for the protocol is widespread in media players, web browsers, mobile devices, and streaming media servers. , an annual video industry survey has consistently found it to be the most popular streaming format. HLS resembles MPEG-DASH in that 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. Based on standard HTTP transactions, HTTP Live Streaming can traverse any firewall or proxy server that lets through standard HTTP traffic, unlike UDP-based protocols such as RTP. This also allows content to be offered from conventional HTTP servers and delivered over widely available HTTP ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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OBS Studio
OBS Studio (also Open Broadcaster Software or OBS, for short) is a free and open-source, cross-platform screencasting and live streaming app. It is available for Windows, macOS, Linux distributions, and BSD. The OBS Project raises funds on Open Collective and Patreon. Overview OBS Studio is a free and open-source app for screencasting and live streaming. Written in C/ C++ and built with Qt, OBS Studio provides real-time capture, scene composition, recording, encoding, and broadcasting via Real-Time Messaging Protocol (RTMP), HLS, SRT, RIST or WebRTC. It can stream videos to any RTMP-supporting destination, including YouTube, Twitch, Instagram and Facebook. For video encoding, OBS Studio can use the x264, AOM-AV1, SVT-AV1 transcoder, Intel Quick Sync Video, Nvidia NVENC, AMD Video Coding Engine and VAAPI to encode video streams into the H.264/MPEG-4 AVC or H.265/HEVC formats. It can encode multiple tracks of audio in AAC and Opus format. More experienced users can ch ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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IETF
The Internet Engineering Task Force (IETF) is a standards organization for the Internet standard, Internet and is responsible for the technical standards that make up the Internet protocol suite (TCP/IP). It has no formal membership roster or requirements and all its participants are volunteers. Their work is usually funded by employers or other sponsors. The IETF was initially supported by the federal government of the United States but since 1993 has operated under the auspices of the Internet Society, a non-profit organization with local chapters around the world. Organization There is no membership in the IETF. Anyone can participate by signing up to a working group mailing list, or registering for an IETF meeting. The IETF operates in a bottom-up task creation mode, largely driven by working groups. Each working group normally has appointed two co-chairs (occasionally three); a charter that describes its focus; and what it is expected to produce, and when. It is open ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Adobe Media Server
Adobe Media Server (AMS) is a proprietary data and media server from Adobe Systems (originally a Macromedia product). This server works with the Flash Player and HTML5 runtime to create media driven, multiuser RIAs ( Rich Internet Applications). The server uses ActionScript 1, an ECMAScript based scripting language, for server-side logic. Prior to version 2, it was known as Flash Communication Server. Prior to version 5, it was known as Flash Media Server. In February 2019, Adobe Systems Incorporated granteVeriskope Incrights to further develop, resell, and extend distribution of the software product. History On March 16, 2002, Macromedia released Flash Player 6. This version included all the functionality for a yet to be released server called Flash Communication Server MX. Version 1.0 was released on 9 July 2002 and included all the basic features that make up the product, including the NetConnection, SharedObject and NetStream objects. Version 1.5 was released on 27 Ma ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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SCTE-35
SCTE-35 (ANSI/SCTE 35 2013) is a joint ANSI/ Society of Cable Telecommunications Engineers standard that describes the inline insertion of cue tones in mpeg-ts streams. The full standard name is "Digital Program Insertion Cueing Message for Cable." History SCTE-35 was originally used in the US to signal a local ad insertion opportunity in the transport streams, and in Europe to insert local TV programs (e.g. local news transmissions). It is now used to signal all kinds of program and ad events in linear transport streams and in newer ABR delivery formats such as HLS and DASH. Work on splicing GOP compressed video streams was started by SMPTE ST 312M Television - Splice Points for MPEG-2 Transport Streams. This standard had some basic signaling and details on how to perform seamless and near seamless splicing of compressed streams. Although it mainly worked on constant bit rate streams, and in the cable industry the digital feeds were moving to a statistical multiplex format. A co ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Video Encoding
In information theory, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compression reduces bits by identifying and eliminating statistical redundancy. No information is lost in lossless compression. Lossy compression reduces bits by removing unnecessary or less important information. Typically, a device that performs data compression is referred to as an encoder, and one that performs the reversal of the process (decompression) as a decoder. The process of reducing the size of a data file is often referred to as data compression. In the context of data transmission, it is called source coding: encoding is done at the source of the data before it is stored or transmitted. Source coding should not be confused with channel coding, for error detection and correction or line coding, the means for mapping data onto a signal. ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Video Quality
Video quality is a characteristic of a video passed through a video transmission or processing system that describes perceived video degradation (typically compared to the original video). Video processing systems may introduce some amount of distortion or artifacts in the video signal that negatively impact the user's perception of the system. For many stakeholders in video production and distribution, ensuring video quality is an important task. Video quality evaluation is performed to describe the quality of a set of video sequences under study. Video quality can be evaluated objectively (by mathematical models) or subjectively (by asking users for their rating). Also, the quality of a system can be determined offline (i.e., in a laboratory setting for developing new codecs or services) or in-service (to monitor and ensure a certain level of quality). From analog to digital video Since the world's first video sequence was recorded and transmitted, many video processing systems ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Streaming
Streaming media refers to multimedia delivered through a network for playback using a media player. Media is transferred in a ''stream'' of packets from a server to a client and is rendered in real-time; this contrasts with file downloading, a process in which the end-user obtains an entire media file before consuming the content. Streaming is more commonly used for video on demand, streaming television, and music streaming services over the Internet. While streaming is most commonly associated with multimedia from a remote server over the Internet, it also includes offline multimedia between devices on a local area network. For example, using DLNA and a home server, or in a personal area network between two devices using Bluetooth (which uses radio waves rather than IP). Online streaming was initially popularized by RealNetworks and Microsoft in the 1990s and has since grown to become the globally most popular method for consuming music and videos, with nume ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MPEG-4 Part 14
MP4 (formally MPEG-4 Part 14), is a digital multimedia container format most commonly used to store video and audio, but it can also be used to store other data such as subtitles and still images. Like most modern container formats, it allows streaming over the Internet. The only filename extension for MPEG-4 Part 14 files as defined by the specification is .mp4. MPEG-4 Part 14 is a standard specified as a part of the MPEG-4 specifications, formally as ISO/ IEC 14496-14:2003. Unlike the audio-only compression formats MP3 and MP2, MP4 is a container format that can hold various types of media from various codecs. During the 2000s, portable media players were sometimes erroneously advertised as "MP4 players", even if they may play a different format like AMV video and not necessarily the MPEG-4 Part 14 format. Data streams Most kinds of data can be embedded in MPEG-4 Part 14 files through ''private streams''. A separate hint track is used to include streaming information in ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MPEG Transport Stream
MPEG transport stream (MPEG-TS, MTS) or simply transport stream (TS) is a standard digital container format for transmission and storage of audio, video, and Program and System Information Protocol (PSIP) data. It is used in broadcast systems such as DVB, ATSC and IPTV. Transport stream specifies a container format encapsulating packetized elementary streams, with error correction and synchronization pattern features for maintaining transmission integrity when the communication channel carrying the stream is degraded. Transport streams differ from the similarly named MPEG program stream in several important ways: program streams are designed for reasonably reliable media, such as discs (like DVDs), while transport streams are designed for less reliable transmission, namely terrestrial or satellite broadcast. Further, a transport stream may carry multiple programs. Transport stream is specified in ''MPEG-2 Part 1, Systems'', formally known as '' ISO/IEC standard 1 ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Dolby Digital Plus
Dolby Digital Plus, also known as Enhanced AC-3 (and commonly abbreviated as DDP, DD+, E-AC-3 or EC-3), is a digital audio compression scheme developed by Dolby Labs for the transport and storage of multi-channel digital audio. It is a successor to Dolby Digital (AC-3), and has a number of improvements over that codec, including support for a wider range of data rates (32 kbit/s to 6144 kbit/s), an increased channel count, and multi-program support (via substreams), as well as additional tools (algorithms) for representing compressed data and counteracting artifacts. Whereas Dolby Digital (AC-3) supports up to five full-bandwidth audio channels at a maximum bitrate of 640 kbit/s, E-AC-3 supports up to 15 full-bandwidth audio channels at a maximum bitrate of 6.144 Mbit/s. The full set of technical specifications for E-AC-3 (and AC-3) are standardized and published in Annex E of ATSC A/52:2012, as well as Annex E of ETSI TS 102 366. Technical details Specif ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Dolby AC-3
Dolby Digital, originally synonymous with Dolby AC-3 (see below), is the name for a family of audio compression technologies developed by Dolby Laboratories. Called Dolby Stereo Digital until 1995, it is lossy compression (except for Dolby TrueHD). The first use of Dolby Digital was to provide digital sound in cinemas from 35 mm film prints. It has since also been used for TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles. Dolby AC-3 was the original version of the Dolby Digital codec. The basis of the Dolby AC-3 multi-channel audio coding standard is the modified discrete cosine transform (MDCT), a lossy audio compression algorithm. It is a modification of the discrete cosine transform (DCT) algorithm, which was proposed by Nasir Ahmed in 1972 for image compression. The DCT was adapted into the MDCT by J.P. Princen, A.W. Johnson and Alan B. Bradley at the University of Surrey in 1987. Dolby Laborator ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |