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Ultra Low Delay Audio Coder
The Ultra Low Delay Audio Coder (ULD) is a development of the Fraunhofer Institute for Digital Media Technology (IDMT), which is headed by one of the fathers of MP3, Karlheinz Brandenburg, and of the Fraunhofer Institute for Integrated Circuits (IIS). The ULD is a lossy audio data compression scheme that only introduces a very small amount of delay into the audio signal compared to commonly known audio coders like MP3 or Advanced Audio Coding, AAC. This property is especially useful for communication purposes (like voice calls, video conferencing or making music via the internet), for which not only high compression ratios are necessary, but low latency is critical, too. External linksHomepage of the ULD Codec [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Karlheinz Brandenburg
Karlheinz Brandenburg (born 20 June 1954) is a German electrical engineer and mathematician. Together with Ernst Eberlein, Heinz Gerhäuser (former Institutes Director of Fraunhofer IIS), Bernhard Grill, Jürgen Herre and Harald Popp (all Fraunhofer IIS), he developed the widespread MP3 method for audio data compression. He is also known for his elementary work in the field of audio coding, the perception measurement, the wave field synthesis and psychoacoustics. Brandenburg has received numerous national and international research awards, prizes and honors for his work. Since 2000 he has been a professor of electronic media technology at the Technical University Ilmenau. Brandenburg was significantly involved in the founding of the Fraunhofer Institute for Digital Media Technology (IDMT) and currently serves as its director. Brandenburg has been called the "father of the MP3" format. Biography Brandenburg received a Dipl. Ing. degree from Erlangen University in Electrical ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Lossy
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 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 used lossy compression al ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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. ... [...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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
