IP codec

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IP codecs are used to send video or audio signals over an IP network such as the Internet. The initials "IP" here stand for "Internet Protocol", while the term "codec" is short for "encoder/decoder" or "compressor/decompressor".

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IP video codecs

IP video codecs are used widely in security and broadcast applications to send video between two locations. Video codecs use compression algorithms to send good video quality at substantially lower bit rates than uncompressed signals. Broadcast applications often use MPEG-2 and H.264/MPEG-4 AVC standards for video compression. The EBU is working on a minimum set of common standards for real-time video over IP transmissions. [1] The recommended standards and protocols are designed to ensure compatibility between different codecs and provide adequate high-quality transmissions.

IP audio codecs

IP audio codecs are used to send broadcast quality audio over IP from remote broadcast locations to radio and television studios around the globe. IP codecs are ideal for use in remote broadcasts, as studio/transmitter links (STLs) or for studio-to-studio audio distribution.

IP audio codecs use audio compression algorithms to send high-fidelity audio over both wired broadband IP networks and wireless broadband networks.

Related Research Articles

A codec is a device or computer program that encodes or decodes a data stream or signal. Codec is a portmanteau of coder/decoder.

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.

<span class="mw-page-title-main">Digital video</span> Digital electronic representation of moving visual images

Digital video is an electronic representation of moving visual images (video) in the form of encoded digital data. This is in contrast to analog video, which represents moving visual images in the form of analog signals. Digital video comprises a series of digital images displayed in rapid succession, usually at 24, 25, 30, or 60 frames per second. Digital video has many advantages such as easy copying, multicasting, sharing and storage.

<span class="mw-page-title-main">ISDN</span> Set of digital telephony standards

Integrated Services Digital Network (ISDN) is a set of communication standards for simultaneous digital transmission of voice, video, data, and other network services over the digitalised circuits of the public switched telephone network. Work on the standard began in 1980 at Bell Labs and was formally standardized in 1988 in the CCITT "Red Book". By the time the standard was released, newer networking systems with much greater speeds were available, and ISDN saw relatively little uptake in the wider market. One estimate suggests ISDN use peaked at a worldwide total of 25 million subscribers at a time when 1.3 billion analog lines were in use. ISDN has largely been replaced with digital subscriber line (DSL) systems of much higher performance.

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.

Speech coding is an application of data compression to digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.

Telephony is the field of technology involving the development, application, and deployment of telecommunications services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is intimately linked to the invention and development of the telephone.

<span class="mw-page-title-main">Digital audio</span> Technology that records, stores, and reproduces sound

Digital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit sample depth. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced analog audio technology in many areas of audio engineering, record production and telecommunications in the 1990s and 2000s.

<span class="mw-page-title-main">Video codec</span> Digital video processing

A video codec is software or hardware that compresses and decompresses digital video. In the context of video compression, codec is a portmanteau of encoder and decoder, while a device that only compresses is typically called an encoder, and one that only decompresses is a decoder.

Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls for the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.

MPEG-1 Audio Layer II or MPEG-2 Audio Layer II is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting.

<span class="mw-page-title-main">DVB-S2</span> Digital satellite television standard

Digital Video Broadcasting - Satellite - Second Generation (DVB-S2) is a digital television broadcast standard that has been designed as a successor for the popular DVB-S system. It was developed in 2003 by the Digital Video Broadcasting Project, an international industry consortium, and ratified by ETSI in March 2005. The standard is based on, and improves upon DVB-S and the electronic news-gathering system, used by mobile units for sending sounds and images from remote locations worldwide back to their home television stations.

<span class="mw-page-title-main">Telephone hybrid</span> Telephone circuit element

In analog telephony, a telephone hybrid is the component at the ends of a subscriber line of the public switched telephone network (PSTN) that converts between two-wire and four-wire forms of bidirectional audio paths. When used in broadcast facilities to enable the airing of telephone callers, the broadcast-quality telephone hybrid is known as a broadcast telephone hybrid or telephone balance unit.

<span class="mw-page-title-main">Video server</span> Device that is dedicated to delivering video

A video server is a computer-based device that is dedicated to delivering video. Video servers are used in a number of applications, and often have additional functions and capabilities that address the needs of particular applications. For example, video servers used in security, surveillance and inspection applications typically are designed to capture video from one or more cameras and deliver the video via a computer network. In video production and broadcast applications, a video server may have the ability to record and play recorded video, and to deliver many video streams simultaneously.

<span class="mw-page-title-main">Internet Protocol television</span> Television transmitted over a computer network

Internet Protocol television (IPTV), also called TV over broadband, is the service delivery of television over Internet Protocol (IP) networks. Usually sold and run by a telecom provider, it consists of broadcast live television that is streamed over the Internet (multicast) — in contrast to delivery through traditional terrestrial, satellite, and cable transmission formats — as well as video on demand services for watching or replaying content (unicast).

<span class="mw-page-title-main">G.722</span> ITU-T recommendation

G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on the same technology is G.726.

<span class="mw-page-title-main">1080p</span> Video mode

1080p is a set of HDTV high-definition video modes characterized by 1,920 pixels displayed across the screen horizontally and 1,080 pixels down the screen vertically; the p stands for progressive scan, i.e. non-interlaced. The term usually assumes a widescreen aspect ratio of 16:9, implying a resolution of 2.1 megapixels. It is often marketed as Full HD or FHD, to contrast 1080p with 720p resolution screens. Although 1080p is sometimes referred to as 2K resolution, other sources differentiate between 1080p and (true) 2K resolution.

A POTS codec is a type of audio coder-decoder (codec) that uses digital signal processing to transmit audio digitally over standard telephone lines at a higher level of audio quality than the telephone line would normally provide in its analog mode. The POTS codec is one of a family of broadcast codecs differentiated by the type of telecommunications circuit used for transmission. The ISDN codec, which instead uses ISDN lines, and the IP codec which uses private or public IP networks are also common.

Audio over IP (AoIP) is the distribution of digital audio across an IP network such as the Internet. It is used increasingly to provide high-quality audio feeds over long distances. The application is also known as audio contribution over IP (ACIP) in reference to the programming contributions made by field reporters and remote events. Audio quality and latency are key issues for contribution links. In the past, these links have made use of ISDN services but these have become increasingly difficult or expensive to obtain.

References

  1. "EBU N/VCIP". Archived from the original on 2010-10-03. Retrieved 2009-07-16.

Further reading