LAME is a software encoder that converts a digitized WAV audio file into the MP3 audio coding file format. LAME is a free software project that was first released in 1998, and has incorporated many improvements since then, including an improved psychoacoustic model. The LAME encoder outperforms early encoders like L3enc.
MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany, with support from other digital scientists in the US and elsewhere. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended — defining additional bit-rates and support for more audio channels — as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG 2.5 — extended to better support lower bit rates — is commonly implemented, but is not a recognized standard.
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.
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. 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.
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting.
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 at the same bit rate.
Surround sound is a technique for enriching the fidelity and depth of sound reproduction by using multiple audio channels from speakers that surround the listener. Its first application was in movie theaters. Prior to surround sound, theater sound systems commonly had three "screen channels" of sound that played from three loudspeakers located in front of the audience. Surround sound adds one or more channels from loudspeakers to the side or behind the listener that are able to create the sensation of sound coming from any horizontal direction around the listener.
Adobe Audition is a digital audio workstation developed by Adobe Inc. featuring both a multitrack, non-destructive mix/edit environment and a destructive-approach waveform editing view.
MP3 Surround is an extension of MP3 for multi-channel audio support including 5.1 surround sound. It was developed by Fraunhofer IIS in collaboration with Thomson and Agere Systems, and released in December 2004.
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.
DTS, Inc. is an American company that makes multichannel audio technologies for film and video. Based in Calabasas, California, the company introduced its DTS technology in 1993 as a competitor to Dolby Laboratories, incorporating DTS in the film Jurassic Park (1993). The DTS product is used in surround sound formats for both commercial/theatrical and consumer-grade applications. It was known as The Digital Experience until 1995. DTS licenses its technologies to consumer electronics manufacturers.
Fraunhofer l3enc was the first public software able to encode PCM (.wav) files to the MP3 format. The first public version was released on July 13, 1994. This commandline tool was shareware and limited to 112 kbit/s. It was available for MS-DOS, Linux, Solaris, SunOS, NeXTstep and IRIX. A licence that allowed full use cost 350 Deutsche Mark, or about $250 (US).
WinPlay3 was the first real-time MP3 audio player for PCs running Windows, both 16-bit and 32-bit. Prior to this, audio compressed with MP3 had to be decompressed prior to listening. It was released by Fraunhofer IIS, creators of the MP3 format, on September 9, 1995. The latest version was released on May 23, 1997. Since then, the Fraunhofer Society has removed any trace and mention of WinPlay3 from their web sites. However, the software remains available by utilizing the Web Archive, or one of the below links.
Ensonido is a real-time post processing algorithm that allows users to play back MP3 Surround files in standard headphones. Ensonido was developed by the Fraunhofer Society. It simulates the natural reception of surround sound by the human ear, which usually receives tones from surrounding loudspeakers and from reflections and echoes of the listening room. The out-of-head localization achieved that way increases the listening comfort noticeably in contrast to conventional stereo headphone listening with its in-head localization of all sounds. In version 3.0 of the Fraunhofer IIS MP3 Surround Player, Ensonido is replaced with newer mp3HD
MPEG-4 SLS, or MPEG-4 Scalable to Lossless as per ISO/IEC 14496-3:2005/Amd 3:2006 (Scalable Lossless Coding), is an extension to the MPEG-4 Part 3 (MPEG-4 Audio) standard to allow lossless audio compression scalable to lossy MPEG-4 General Audio coding methods (e.g., variations of AAC). It was developed jointly by the Institute for Infocomm Research (I2R) and Fraunhofer, which commercializes its implementation of a limited subset of the standard under the name of HD-AAC. Standardization of the HD-AAC profile for MPEG-4 Audio is under development (as of September 2009).
MPEG Surround, also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo audio services to multi-channel audio in a backwards compatible fashion. The total bit rates used for the core and the MPEG Surround data are typically only slightly higher than the bit rates used for coding of the core. MPEG Surround adds a side-information stream to the core bit stream, containing spatial image data. Legacy stereo playback systems will ignore this side-information while players supporting MPEG Surround decoding will output the reconstructed multi-channel audio.
In audio engineering, joint encoding refers to a joining of several channels of similar information during encoding in order to obtain higher quality, a smaller file size, or both.
Unified Speech and Audio Coding (USAC) is an audio compression format and codec for both music and speech or any mix of speech and audio using very low bit rates between 12 and 64 kbit/s. It was developed by Moving Picture Experts Group (MPEG) and was published as an international standard ISO/IEC 23003-3 and also as an MPEG-4 Audio Object Type in ISO/IEC 14496-3:2009/Amd 3 in 2012.
Fraunhofer FDK AAC is an open-source software library for encoding and decoding Advanced Audio Coding (AAC) format audio, developed by Fraunhofer IIS, and included as part of Android 4.1. It supports several Audio Object Types including MPEG-2 and MPEG-4 AAC LC, HE-AAC, HE-AACv2 as well AAC-LD and AAC-ELD for real-time communication. The encoding library supports sample rates up to 96 kHz and up to eight channels.
MPEG-H 3D Audio, specified as ISO/IEC 23008-3, 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.
