|
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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Plain Old Telephone Service
Plain old telephone service (POTS), or plain ordinary telephone system, is a retronym for voice-grade telephone service employing analog signal transmission over copper loops. POTS was the standard service offering from telephone companies from 1876 until 1988 in the United States when the Integrated Services Digital Network (ISDN) Basic Rate Interface (BRI) was introduced, followed by cellular telephone systems, and voice over IP (VoIP). POTS remains the basic form of residential and small business service connection to the telephone network in many parts of the world. The term reflects the technology that has been available since the introduction of the public telephone system in the late 19th century, in a form mostly unchanged despite the introduction of Touch-Tone dialing, electronic telephone exchanges and fiber-optic communication into the public switched telephone network (PSTN). Characteristics POTS is characterized by several aspects: *Bi-directional (full duplex) comm ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Mixing Console
A mixing console or mixing desk is an electronic device for mixing audio signals, used in sound recording and reproduction and sound reinforcement systems. Inputs to the console include microphones, signals from electric or electronic instruments, or recorded sounds. Mixers may control analog or digital signals. The modified signals are summed to produce the combined output signals, which can then be broadcast, amplified through a sound reinforcement system or recorded. Mixing consoles are used for applications including recording studios, public address systems, sound reinforcement systems, nightclubs, broadcasting, and post-production. A typical, simple application combines signals from microphones on stage into an amplifier that drives one set of loudspeakers for the audience. A DJ mixer may have only two channels, for mixing two record players. A coffeehouse's tiny stage might only have a six-channel mixer, enough for two singer-guitarists and a percussionist. A nigh ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Tieline
Tieline Technology has offices in Indianapolis in the United States (Tieline America LLC) and in Perth, Western Australia (Tieline Pty Ltd). The company has a wide and established distribution network throughout Europe, the Americas and Australasia. Tieline develops a range of broadcast audio codecs that are sold to television and radio networks around the globe. All Tieline codecs are IP codecs, ISDN codecs, POTS codecs, GSM codecs, X.21 codecs and satellite-capable (IP and ISDN) codecs. Broadcasters use these codecs for remote broadcasts (outside broadcasting), for audio distribution between studios and for studio/transmitter link (STL) applications. Tieline codecs are Session Initiation Protocol (SIP) compatible and Tieline and ten other codec manufacturers have successfully tested IP Interoperability using SIP to connect according to EBU N/ACIP tech 3326 specifications relating to sending audio over IP. History The company was founded in 1981 by John Gouteff and Rod Henderso ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Comrex
Comrex is an American corporation that designs and manufactures equipment for radio and television broadcasting. Beginnings Comrex was founded in 1961 by John Cheney, a broadcast engineer. His mission, as outlined in Comrex's inaugural press release, is "to apply advanced state of the art knowledge and techniques to the production of high quality, practical equipment which can be operated by non-technical personnel.” Throughout the 1960s and 1970s, Comrex developed audio products for the television market: * 1968 – Cheney develops Models 7035 and 7040, receivers for wireless microphone systems. * 1973 – The 450 RA/TA wireless microphone system, designed for TV news, is the first to enable reporters to move more than 1000 feet from the camera. * 1975 – The Comrex wireless cue system, composed of the CTA cue transmitter and the CRA (now LPQRA) receiver, enables field reporters and producers to hear production audio and instructions, without a wired connection to a news ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Bandwidth (computing)
In computing, bandwidth is the maximum rate of data transfer across a given path. Bandwidth may be characterized as network bandwidth, data bandwidth, or digital bandwidth. This definition of ''bandwidth'' is in contrast to the field of signal processing, wireless communications, modem data transmission, digital communications, and electronics, in which ''bandwidth'' is used to refer to analog signal bandwidth measured in hertz, meaning the frequency range between lowest and highest attainable frequency while meeting a well-defined impairment level in signal power. The actual bit rate that can be achieved depends not only on the signal bandwidth but also on the noise on the channel. Network capacity The term ''bandwidth'' sometimes defines the net bit rate 'peak bit rate', 'information rate,' or physical layer 'useful bit rate', channel capacity, or the maximum throughput of a logical or physical communication path in a digital communication system. For example, bandwidth test ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Fault Tolerance
Fault tolerance is the property that enables a system to continue operating properly in the event of the failure of one or more faults within some of its components. If its operating quality decreases at all, the decrease is proportional to the severity of the failure, as compared to a naively designed system, in which even a small failure can cause total breakdown. Fault tolerance is particularly sought after in high-availability, mission-critical, or even life-critical systems. The ability of maintaining functionality when portions of a system break down is referred to as graceful degradation. A fault-tolerant design enables a system to continue its intended operation, possibly at a reduced level, rather than failing completely, when some part of the system fails. The term is most commonly used to describe computer systems designed to continue more or less fully operational with, perhaps, a reduction in throughput or an increase in response time in the event of some partial fa ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Redundancy (information Theory)
In information theory, redundancy measures the fractional difference between the entropy of an ensemble , and its maximum possible value \log(, \mathcal_X, ). Informally, it is the amount of wasted "space" used to transmit certain data. Data compression is a way to reduce or eliminate unwanted redundancy, while forward error correction is a way of adding desired redundancy for purposes of error detection and correction when communicating over a noisy channel of limited capacity. Quantitative definition In describing the redundancy of raw data, the rate of a source of information is the average entropy per symbol. For memoryless sources, this is merely the entropy of each symbol, while, in the most general case of a stochastic process, it is :r = \lim_ \frac H(M_1, M_2, \dots M_n), in the limit, as ''n'' goes to infinity, of the joint entropy of the first ''n'' symbols divided by ''n''. It is common in information theory to speak of the "rate" or "entropy" of a language. Th ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Variable Bitrate
Variable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding. As opposed to constant bitrate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bitrate (and therefore more storage space) to be allocated to the more complex segments of media files while less space is allocated to less complex segments. The average of these rates can be calculated to produce an average bitrate for the file. MP3, WMA and AAC audio files can optionally be encoded in VBR, while Opus and Vorbis are encoded in VBR by default. Variable bit rate encoding is also commonly used on MPEG-2 video, MPEG-4 Part 2 video (Xvid, DivX, etc.), MPEG-4 Part 10/H.264 video, Theora, Dirac and other video compression formats. Additionally, variable rate encoding is inherent in lossless compression schemes such as FLAC and Apple Lossless. Advantages and disadvantages of VBR The advantages of VBR are that it pr ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Latency (engineering)
Latency, from a general point of view, is a time delay between the cause and the effect of some physical change in the system being observed. Lag, as it is known in gaming circles, refers to the latency between the input to a simulation and the visual or auditory response, often occurring because of network delay in online games. Latency is physically a consequence of the limited velocity at which any physical interaction can propagate. The magnitude of this velocity is always less than or equal to the speed of light. Therefore, every physical system with any physical separation (distance) between cause and effect will experience some sort of latency, regardless of the nature of the stimulation at which it has been exposed to. The precise definition of latency depends on the system being observed or the nature of the simulation. In communications, the lower limit of latency is determined by the medium being used to transfer information. In reliable two-way communication syst ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
AAC-LD
The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) 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 (ISO/IEC 14496-3:1999/Amd 1:2000) and in its later revisions. AAC-LD uses a version of the modified discrete cosine transform (MDCT) audio coding technique called the LD-MDCT. AAC-LD is widely used by Apple as the voice-over-IP (VoIP) speech codec in FaceTime. Real time CODEC requirements The most stringent requirements are a maximum algorithmic delay of only 20 ms and a good audio quality for all kind of audio signals including speech and music. * The AAC-LD coding scheme bridges the gap between speech coding schemes and high quality audio coding schemes. Two-way communication with AAC-LD is possible on usual analog telephone lines and via ISD ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Apt-X
aptX (''apt'' stands for ''audio processing technology'') is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications. History The original aptX algorithm was developed in the 1980s by Dr. Stephen Smyth as part of his Ph.D. research at Queen's University Belfast School of Electronics, Electrical Engineering and Computer Science; its design is based on time domain ADPCM principles without psychoacoustic auditory masking techniques. aptX audio coding was first introduced to the commercial market as a semiconductor product, a custom programmed DSP integrated circuit with part name APTX100ED, which was initially adopted by broadcast automation equipment manufacturers who required a means to store CD-quality audio on a computer hard disk drive for automatic playout during a radio show, for example, hence replacing the task of the disc jockey. The company was bought by Solid State Logic ca. 1988, and became pa ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
.jpg "Click for more on -> Mixing Console")
