Telephone Hybrid
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Telephone Hybrid
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. The need for hybrids comes from the nature of analog plain old telephone service (POTS) home or small business telephone lines, where the two audio directions are combined on a single two-wire pair. Within the telephone network, switching and transmission are almost always four-wire circuits with the two signals being separated. Hybrids perform the necessary conversion. In older analog networks, conversion to four-wire was required so that repeater amplifiers could be inserted in long-distance links. In today's digital systems, each speech direction must be processed and transported independently. The line ...
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Telephone Hybrid Transformer
A telephone is a telecommunications device that permits two or more users to conduct a conversation when they are too far apart to be easily heard directly. A telephone converts sound, typically and most efficiently the human voice, into electronic signals that are transmitted via cables and other communication channels to another telephone which reproduces the sound to the receiving user. The term is derived from el, τῆλε (''tēle'', ''far'') and φωνή (''phōnē'', ''voice''), together meaning ''distant voice''. A common short form of the term is ''phone'', which came into use early in the telephone's history. In 1876, Alexander Graham Bell was the first to be granted a United States patent for a device that produced clearly intelligible replication of the human voice at a second device. This instrument was further developed by many others, and became rapidly indispensable in business, government, and in households. The essential elements of a telephone are a micr ...
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Automatic Gain Control
Automatic gain control (AGC) is a closed-loop feedback regulating circuit in an amplifier or chain of amplifiers, the purpose of which is to maintain a suitable signal amplitude at its output, despite variation of the signal amplitude at the input. The average or peak output signal level is used to dynamically adjust the gain of the amplifiers, enabling the circuit to work satisfactorily with a greater range of input signal levels. It is used in most radio receivers to equalize the average volume (loudness) of different radio stations due to differences in received signal strength, as well as variations in a single station's radio signal due to fading. Without AGC the sound emitted from an AM radio receiver would vary to an extreme extent from a weak to a strong signal; the AGC effectively reduces the volume if the signal is strong and raises it when it is weaker. In a typical receiver the AGC feedback control signal is usually taken from the detector stage and applied to ...
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Advanced Audio Coding
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves higher sound quality than MP3 encoders at the same bit rate. AAC has been standardized by ISO and IEC as part of the MPEG-2 and MPEG-4 specifications.ISO (2006ISO/IEC 13818-7:2006 - Information technology -- Generic coding of moving pictures and associated audio information -- Part 7: Advanced Audio Coding (AAC), Retrieved on 2009-08-06ISO (2006, Retrieved on 2009-08-06 Part of AAC, HE-AAC ("AAC+"), is part of MPEG-4 Audio and is adopted into digital radio standards DAB+ and Digital Radio Mondiale, and mobile television standards DVB-H and ATSC-M/H. AAC supports inclusion of 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low frequency effects ( LFE, limited to 120 Hz) channels, up to 16 "coupling" or dialog channels, and up to 16 data streams. The quality for stereo is satisf ...
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Audio Data Compression
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 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. C ...
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Remote Broadcast
In broadcast engineering, a remote broadcast (usually just called a remote or a live remote, or in news parlance, a live shot) is broadcasting done from a location away from a formal television 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 television station, where it joins the normal airchain. Other methods include satellite trucks, production trucks and even regular telephone lines if necessary. History The first airing of a remote broadcast came in 1924, when Loew's Theater publicist and WHN (New York City) station manager Nils Granlund leased telegraph lines from Western Union to provide the first link in what became called cabaret broadcasting." By early 1925, Granlund had established remote lines between WHN and more than thirty New York City jazz nightclubs, including the Silver Slipper, The Parody Club, the Cotton Club, the Strand Roof, and Club Moritz. These big ban ...
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POTS Codec
A POTS codec is a type of audio coder-decoder (codec) that uses digital signal processing to transmit audio digitally over standard telephone lines ("Plain Old Telephone Service") 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. Primarily used in broadcast engineering to link remote broadcast locations to the host studio, a hardware codec, implemented with digital signal processing, is used to compress the audio data enough to travel through a pair of a 33.6k modems. POTS codecs have the disadvantages of being restricted to relatively low bit rates and being susceptible to variable line quality. ISDN and IP codecs have the advantage of being natively digital, and operate a ...
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Radio Producer
A radio producer oversees the making of a radio show. The job title covers several different job descriptions: *Content producers or executive producers oversee and orchestrate a radio show or feature. The content producer might organize music choices, guests and callers for a talk radio show or competitions, timings and overall show content. *Creative producers, imaging specialists or imaging producers ("Imaging" in the radio business relates to brand management rather than visual material) produce audio content for the show, such as sound clips used on the show and "promo clips" (radio commercials and commercial bumpers used on the same station to promote the show). *The board operator or technical operator operates the technical controls—such as sound volume levels, recording software and switchboard. The producer can be in a separate control room separated from the radio studio by a window, which allowed visual contact while blocking noise. With quieter switches and better ...
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Mix-minus
In audio engineering, a mix-minus or clean feed is a particular setup of a mixing console or matrix mixer, such that an output of the mixer contains everything ''except'' a designated input. Mix-minus is often used to prevent echoes or feedback in broadcast or sound reinforcement systems. Examples The most common situation in which a mix-minus is used is when a telephone hybrid is connected to a console, usually at a radio station. The person on the telephone hears all relevant feeds, usually an identical mix to that of the master bus, including the DJ's mic feed, except that the caller does not hear their own voice. Because the feed being returned to the telephone has an identical mix, minus only one input, it is most efficient to employ a mix-minus as opposed to creating and mixing for a new independent output bus. Mix-minus is also often used together with IFB systems in electronic news gathering (ENG) for television news reporters and interview subjects speaking to a ho ...
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Caller ID
Caller identification (Caller ID) is a telephone service, available in analog and digital telephone systems, including voice over IP (VoIP), that transmits a caller's telephone number to the called party's telephone equipment when the call is being set up. The caller ID service may include the transmission of a name associated with the calling telephone number, in a service called Calling Name Presentation (CNAM). The service was first defined in 1993 in International Telecommunication Union—Telecommunication Standardization Sector (ITU-T) Recommendation Q.731.3. The information received from the service is displayed on a telephone display screen, on a separately attached device, or on other displays, such as cable television sets when telephone and television service is provided by the same vendor. Value to society includes use by suicide-prevention hot lines and enabling businesses "like pizza restaurants and florists" to quickly have confidence in telephoned orders. The ...
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DTMF
Dual-tone multi-frequency signaling (DTMF) is a telecommunication signaling system using the voice-frequency band over telephone lines between telephone equipment and other communications devices and switching centers. DTMF was first developed in the Bell System in the United States, and became known under the trademark Touch-Tone for use in push-button telephones supplied to telephone customers, starting in 1963. DTMF is standardized as ITU-T Recommendation Q.23. It is also known in the UK as ''MF4''. The Touch-Tone system using a telephone keypad gradually replaced the use of rotary dial and has become the industry standard for landline and mobile service. Other multi-frequency systems are used for internal signaling within the telephone network. Multifrequency signaling Before the development of DTMF, telephone numbers were dialed by users with a loop-disconnect (LD) signaling, more commonly known as pulse dialing (dial pulse, DP) in the United States. It functions by int ...
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XLR Connector
The XLR connector is a type of electrical connector primarily used in professional audio, video, and stage lighting equipment. XLR connectors are cylindical in design, and have three to seven connector pins, and are often employed for analog balanced audio interconnections, AES3 digital audio, portable intercom, DMX512 lighting control, and for low-voltage power supply. XLR connectors are included to the international standard for dimensions, IEC 61076-2-103. The XLR connector is superficially similar to the smaller DIN connector, with which it is physically incompatible. History and manufacturers The XLR connector (also Cannon plug and Cannon connector) was invented by James H. Cannon, founder of the Cannon Electric company, Los Angeles, California. The XLR connector originated from the ''Cannon X'' series of connectors in 1915; by 1950, a latching mechanism was added to the connector, which produced the ''Cannon XL'' model of connector, and by 1955, the female connector ...
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AES3
AES3 is a standard for the exchange of digital audio signals between professional audio devices. An AES3 signal can carry two channels of pulse-code-modulated digital audio over several transmission media including balanced lines, unbalanced lines, and optical fiber. AES3 was jointly developed by the Audio Engineering Society (AES) and the European Broadcasting Union (EBU) and so is also known as AES/EBU. The standard was first published in 1985 and was revised in 1992 and 2003. AES3 has been incorporated into the International Electrotechnical Commission's standard IEC 60958, and is available in a consumer-grade variant known as S/PDIF. History and development The development of standards for digital audio interconnect for both professional and domestic audio equipment, began in the late 1970s in a joint effort between the Audio Engineering Society and the European Broadcasting Union, and culminated in the publishing of AES3 in 1985. The AES3 standard has been revised in ...
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