A codec is a device or computer program that encodes or decodes a data stream or signal. Codec is a portmanteau of coder/decoder.
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.
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.
Digital subscriber line is a family of technologies that are used to transmit digital data over telephone lines. In telecommunications marketing, the term DSL is widely understood to mean asymmetric digital subscriber line (ADSL), the most commonly installed DSL technology, for Internet access.
Telephony is the field of technology involving the development, application, and deployment of telecommunication 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.
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.
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.
In broadcast engineering, a remote broadcast is broadcasting done from a location away from a formal television or radio studio and is considered an electronic field production (EFP). A remote pickup unit (RPU) is usually used to transmit the audio and/or video back to the broadcast station, where it joins the normal airchain. Other methods include satellite trucks, production trucks and even regular telephone lines if necessary.
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.
Digital access carrier system (DACS) is the name used by British Telecom in the United Kingdom for a 0+2 pair gain system.
H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.
The MPEG-4 Low Delay Audio Coder is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) standard. It was published in MPEG-4 Audio Version 2 and in its later revisions.
Telos Alliance is an American corporation manufacturing audio products primarily for broadcast stations. Headquartered in Cleveland, Ohio, US, the company is divided into six divisions:
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.
Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 100 Hz to 17 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16 bits to encode samples, also resulting in much better voice quality.
Constrained Energy Lapped Transform (CELT) is an open, royalty-free lossy audio compression format and a free software codec with especially low algorithmic delay for use in low-latency audio communication. The algorithms are openly documented and may be used free of software patent restrictions. Development of the format was maintained by the Xiph.Org Foundation and later coordinated by the Opus working group of the Internet Engineering Task Force (IETF).
aptX is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications.
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".
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.
Comrex is an American corporation that designs and manufactures equipment for radio and television broadcasting.
