|
Streaming Radio
Online radio (also web radio, net radio, streaming radio, e-radio, IP radio, Internet radio) is a digital audio service transmitted via the Internet. Broadcasting on the Internet is usually referred to as webcasting since it is not transmitted broadly through wireless means. It can either be used as a stand-alone device running through the Internet, or as a software running through a single computer. Internet radio is generally used to communicate and easily spread messages through the form of talk. It is distributed through a wireless communication network connected to a switch packet network (the internet) via a disclosed source. Internet radio involves streaming media, presenting listeners with a continuous stream of audio that typically cannot be paused or replayed, much like traditional broadcast media; in this respect, it is distinct from on-demand file serving. Internet radio is also distinct from podcasting, which involves downloading rather than streaming. Internet ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
A Kerbango
A, or a, is the first letter and the first vowel of the Latin alphabet, used in the modern English alphabet, the alphabets of other western European languages and others worldwide. Its name in English is ''a'' (pronounced ), plural ''aes''. It is similar in shape to the Ancient Greek letter alpha, from which it derives. The uppercase version consists of the two slanting sides of a triangle, crossed in the middle by a horizontal bar. The lowercase version can be written in two forms: the double-storey a and single-storey ɑ. The latter is commonly used in handwriting and fonts based on it, especially fonts intended to be read by children, and is also found in italic type. In English grammar, " a", and its variant " an", are indefinite articles. History The earliest certain ancestor of "A" is aleph (also written 'aleph), the first letter of the Phoenician alphabet, which consisted entirely of consonants (for that reason, it is also called an abjad to distinguish it fro ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
AacPlus
High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio defined as an MPEG-4 Audio profile in ISO/IEC 14496–3. It is an extension of Low Complexity AAC (AAC-LC) optimized for low-bitrate applications such as streaming audio. The usage profile HE-AAC v1 uses spectral band replication (SBR) to enhance the modified discrete cosine transform (MDCT) compression efficiency in the frequency domain. The usage profile HE-AAC v2 couples SBR with Parametric Stereo (PS) to further enhance the compression efficiency of stereo signals. HE-AAC is used in digital radio standards like HD Radio, DAB+ and Digital Radio Mondiale. History The progenitor of HE-AAC was developed by Coding Technologies by combining MPEG-2 AAC-LC with a proprietary mechanism for spectral band replication (SBR), to be used by XM Radio for their satellite radio service. Subsequently, Coding Technologies submitted their SBR mechanism to MPEG as a basis of ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
HE-AAC
High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio defined as an MPEG-4 Audio profile in ISO/IEC 14496–3. It is an extension of Low Complexity AAC (AAC-LC) optimized for low-bitrate applications such as streaming audio. The usage profile HE-AAC v1 uses spectral band replication (SBR) to enhance the modified discrete cosine transform (MDCT) compression efficiency in the frequency domain. The usage profile HE-AAC v2 couples SBR with Parametric Stereo (PS) to further enhance the compression efficiency of stereo signals. HE-AAC is used in digital radio standards like HD Radio, DAB+ and Digital Radio Mondiale. History The progenitor of HE-AAC was developed by Coding Technologies by combining MPEG-2 AAC-LC with a proprietary mechanism for spectral band replication (SBR), to be used by XM Radio for their satellite radio service. Subsequently, Coding Technologies submitted their SBR mechanism to MPEG as a basis of ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
RealAudio
RealAudio, or also spelled as Real Audio is a proprietary audio format developed by RealNetworks and first released in April 1995. It uses a variety of audio codecs, ranging from low-bitrate formats that can be used over dialup modems, to high-fidelity formats for music. It can also be used as a streaming audio format, that is played at the same time as it is downloaded. In the past, many internet radio stations used RealAudio to stream their programming over the internet in real time. In recent years, however, the format has become less common and has given way to more popular audio formats. RealAudio was heavily used by the BBC websites until 2009, though it was discontinued due to its declining use. BBC World Service, the last of the BBC websites to use RealAudio, discontinued its use in March 2011. File extensions RealAudio files were originally identified by a filename extension of .ra (for Real Audio). In 1997, RealNetworks also began offering a video format called RealVideo ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Windows Media Audio
Windows Media Audio (WMA) is a series of audio codecs and their corresponding audio coding formats developed by Microsoft. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as ''WMA'', was conceived as a competitor to the popular MP3 and RealAudio codecs. ''WMA Pro'', a newer and more advanced codec, supports multichannel and high resolution audio. A lossless codec, ''WMA Lossless'', compresses audio data without loss of audio fidelity (the regular WMA format is lossy). ''WMA Voice'', targeted at voice content, applies compression using a range of low bit rates. Microsoft has also developed a digital container format called Advanced Systems Format to store audio encoded by WMA. Development history The first WMA codec was based on earlier work by Henrique Malvar and his team which was transferred to the Windows Media team at Microsoft. Malvar was a senior researcher and manager ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Opus (audio Format)
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. Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed for maximal efficiency. Bitrate, audio bandwidth, complexity, and algorithm can all be adjusted seamlessly in each frame. Opus has the low algorithmic delay (26.5 ms by default) necessary for use as part of a real-time communication link, networke ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Vorbis
Vorbis is a free and open-source software project headed by the Xiph.Org Foundation. The project produces an audio coding format and software reference encoder/decoder (codec) for lossy audio compression. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis. Vorbis is a continuation of audio compression development started in 1993 by Chris Montgomery. Intensive development began following a September 1998 letter from the Fraunhofer Society announcing plans to charge licensing fees for the MP3 audio format. The Vorbis project started as part of the Xiphophorus company's Ogg project (also known as OggSquish multimedia project). Chris Montgomery began work on the project and was assisted by a growing number of other developers. They continued refining the source code until the Vorbis file format was frozen for 1.0 in May 2000. Originally licensed as LGPL, in 2001 the Vorbis license was changed to the BSD lic ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Codec
An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC). In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contains bo ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Lossy Data Compression
In information technology, lossy compression or irreversible compression is the class of data compression methods that uses inexact approximations and partial data discarding to represent the content. These techniques are used to reduce data size for storing, handling, and transmitting content. The different versions of the photo of the cat on this page show how higher degrees of approximation create coarser images as more details are removed. This is opposed to Lossless compression, lossless data compression (reversible data compression) which does not degrade the data. The amount of data reduction possible using lossy compression is much higher than using lossless techniques. Well-designed lossy compression technology often reduces file sizes significantly before degradation is noticed by the end-user. Even when noticeable by the user, further data reduction may be desirable (e.g., for real-time communication or to reduce transmission times or storage needs). The most widely use ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Streaming Media
Streaming media is multimedia that is delivered and consumed in a continuous manner from a source, with little or no intermediate storage in network elements. ''Streaming'' refers to the delivery method of content, rather than the content itself. Distinguishing delivery method from the media applies specifically to telecommunications networks, as most of the traditional media delivery systems are either inherently ''streaming'' (e.g. radio, television) or inherently ''non-streaming'' (e.g. books, videotape, audio CDs). There are challenges with streaming content on the Internet. For example, users whose Internet connection lacks sufficient bandwidth may experience stops, lags, or poor buffering of the content, and users lacking compatible hardware or software systems may be unable to stream certain content. With the use of buffering of the content for just a few seconds in advance of playback, the quality can be much improved. Livestreaming is the real-time delivery of co ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
_label.svg "Click for more on -> Windows Media Audio")
