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MDCT
The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped: it is designed to be performed on consecutive blocks of a larger dataset, where subsequent blocks are overlapped so that the last half of one block coincides with the first half of the next block. This overlapping, in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications, since it helps to avoid artifacts stemming from the block boundaries. As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook, Advanced Audio Coding (AAC), High-Definition Coding (HDC), LDAC, Dolby AC-4, and MPEG-H 3D Audio, as well as speech coding standards such as AAC- ... [...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] |
ATRAC
Adaptive Transform Acoustic Coding (ATRAC) is a family of proprietary audio compression algorithms developed by Sony. MiniDisc was the first commercial product to incorporate ATRAC in 1992. ATRAC allowed a relatively small disc like MiniDisc to have the same running time as CD while storing audio information with minimal loss in perceptible quality. Improvements to the codec in the form of ATRAC3, ATRAC3plus, and ATRAC Advanced Lossless followed in 1999, 2002, and 2006 respectively. Files in ATRAC3 format originally have the extension, however in most cases the files would be stored in an OpenMG Audio container using the extension . Initially files that were encrypted with OpenMG had the extension, before it was replaced by starting in SonicStage v2.1. Encryption is no longer compulsory as of v3.2. Other MiniDisc manufacturers such as Sharp and Panasonic also implemented their own versions of the ATRAC codec. History ATRAC was developed for Sony's MiniDisc format. ATRAC was ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Coding Standards
An audio coding format (or sometimes audio compression format) is a content representation format for storage or transmission of digital audio (such as in digital television, digital radio and in audio and video files). Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software. Some audio coding formats are documented by a detailed technical specification document known as an audio coding specification. Some such specifications are written and approved by standardization organizations as technical standards, and are thus known as an audio coding standard. The term "standard" is also sometimes used for ''de facto'' standards as well as forma ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Dolby Digital
Dolby Digital, originally synonymous with Dolby AC-3, is the name for what has now become a family of audio compression technologies developed by Dolby Laboratories. Formerly named Dolby Stereo Digital until 1995, the audio compression is lossy (except for Dolby TrueHD), based on the modified discrete cosine transform (MDCT) algorithm. The first use of Dolby Digital was to provide digital sound in cinemas from 35 mm film prints; today, it is also used for applications such as TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles. The main basis of the Dolby AC-3 multi-channel audio coding standard is the modified discrete cosine transform (MDCT), a lossy audio compression algorithm. It is a modification of the discrete cosine transform (DCT) algorithm, which was first proposed by Nasir Ahmed in 1972 and was originally intended for image compression. The DCT was adapted into the modified discrete cosine transform (MD ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speech Coding
Speech coding is an application of data compression of 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. Some applications of speech coding are mobile telephony and voice over IP (VoIP). The most widely used speech coding technique in mobile telephony is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques. The techniques employed in speech coding are similar to those used in audio data compression and audio coding where knowledge in psychoacoustics is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in the frequency band 400 to 3500 Hz is transmitted but the reconst ... [...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. C ... [...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] |
Lossy 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 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] |
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] |
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] |
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] |
MPEG-H 3D Audio
MPEG-H 3D Audio, specified as ISO/IEC 23008-3 (MPEG-H Part 3), is an audio coding standard developed by the ISO/IEC Moving Picture Experts Group (MPEG) to support coding audio as audio channels, audio objects, or higher order ambisonics (HOA). MPEG-H 3D Audio can support up to 64 loudspeaker channels and 128 codec core channels. Objects may be used alone or in combination with channels or HOA components. The use of audio objects allows for interactivity or personalization of a program by adjusting the gain or position of the objects during rendering in the MPEG-H decoder. Audio is encoded using an improved modified discrete cosine transform (MDCT) algorithm. Channels, objects, and HOA components may be used to transmit immersive sound as well as mono, stereo, or surround sound. The MPEG-H 3D Audio decoder renders the bitstream to a number of standard speaker configurations as well as to misplaced speakers. Binaural rendering of sound for headphone listening is also supported. His ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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