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 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.
MP3 is a coding format for digital audio. 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.
Karlheinz Brandenburg is a German electrical engineer and mathematician. Together with Ernst Eberlein, Heinz Gerhäuser, Bernhard Grill, Jürgen Herre and Harald Popp, 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 is 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.
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 better sound quality than MP3 at the same bit rate. The confusingly named AAC+ (HE-AAC) does so only at low bit rates and less so at high ones.
A codec is a device or computer program for encoding or decoding a digital data stream or signal. Codec is a portmanteau of coder-decoder.
In signal processing, data compression, source coding, or bit-rate reduction involves encoding information using fewer bits than the original representation. Compression can be 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.
LAME is a software encoder that converts audio to the MP3 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.
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.
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.
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.
Xiph.Org Foundation is a non-profit organization that produces free multimedia formats and software tools. It focuses on the Ogg family of formats, and the most successful one has been Vorbis, an open and freely licensed audio format and codec designed to compete with the patented WMA, MP3 and AAC. As of 2013, the current development work is focusing on Daala, an open and patent-free video format and codec designed to compete with VP9 and the patented High Efficiency Video Coding.
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 optimized for low-bitrate applications such as streaming audio. HE-AAC version 1 profile uses spectral band replication (SBR) to enhance the compression efficiency in the frequency domain. HE-AAC version 2 profile couples SBR with Parametric Stereo (PS) to enhance the compression efficiency of stereo signals. It is a standardized and improved version of the AACplus codec.
FAAC or Freeware Advanced Audio Coder is a software project which includes the AAC encoder FAAC and decoder FAAD2. It supports MPEG-2 AAC as well as MPEG-4 AAC. It supports several MPEG-4 Audio object types, file formats, multichannel and gapless encoding/decoding and MP4 metadata tags. The encoder and decoder is compatible with standard-compliant audio applications using one or more of these object types and facilities. It also supports Digital Radio Mondiale.
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).
The MPEG-4 Low Delay Audio Coder 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 and in its later revisions.
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.
Constrained Energy Lapped Transform (CELT) is an open, royalty-free lossy audio compression format and a free software codec with especially low algorithmic delay for use in low-latency audio communication. The algorithms are openly documented and may be used free of software patent restrictions. Development of the format was maintained by the Xiph.Org Foundation and later coordinated by the Opus working group of the Internet Engineering Task Force (IETF).
In digital audio data reduction technology, aptX is a family of proprietary audio codec compression algorithms owned by Qualcomm.
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. 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.
Harald Popp is a German electrical engineer. Together with Karlheinz Brandenburg, Ernst Eberlein, Heinz Gerhäuser, Bernhard Grill, Jürgen Herre, he developed the widespread MP3 method for audio data compression.