199 relations: A-law algorithm, AAC-LD, Adaptive differential pulse-code modulation, Adaptive Multi-Rate audio codec, Adaptive Multi-Rate Wideband, Adaptive Transform Acoustic Coding, Adobe Flash, Adobe Flash Player, Advanced Audio Coding, Algebraic code-excited linear prediction, Algorithm, AMSN, Android (operating system), Apple Inc., Apple Lossless, AptX, Asao (codec), Asterisk (PBX), ATSC standards, Audio codec, Audio coding format, Audio Lossless Coding, Berkeley Software Distribution, Bit rate, Bluetooth, Bluetooth Special Interest Group, Broadcom Inc., Canadian Electroacoustic Community, CELT, Chromium OS, Cisco Systems, CLIÉ, Code-excited linear prediction, Codec 2, Codec listening test, Companding, Comparison of audio player software, Comparison of video codecs, Comparison of video container formats, Comparison of video player software, Constant bitrate, Cook Codec, Core Audio, Digital audio broadcasting, Digital container format, Digital rights management, Digital television, Digital Video Broadcasting, DirectShow, Dolby Atmos, ..., Dolby Digital, Dolby Digital Plus, Dolby Laboratories, Dolby TrueHD, DTS (sound system), DTS-HD Master Audio, DVD, DVD player, Ekiga, Encoder, Enhanced full rate, ETSI, Extended Adaptive Multi-Rate – Wideband, FAAC, FairPlay, Ffdshow, FFmpeg, FLAC, Flip4Mac, Fraunhofer FDK AAC, Fraunhofer Society, FreeSWITCH, Full Rate, G.711, G.718, G.719, G.722, G.722.1, G.723, G.723.1, G.726, G.728, G.729, G.729.1, GStreamer, Half Rate, Harmonic Vector Excitation Coding, Harris Corporation, High-Efficiency Advanced Audio Coding, International Electrotechnical Commission, International Organization for Standardization, Internet Engineering Task Force, Internet Low Bitrate Codec, Internet Speech Audio Codec, IOS, ISO base media file format, ITU-T, ITunes, L3enc, LAME, Latency (audio), Linear prediction, Linear predictive coding, Linux, List of codecs, List of open-source codecs, Logarithmic scale, Long-term prediction (communications), Lossless compression, Lossless predictive audio compression, Lossy compression, MacOS, Media Foundation, Meridian Lossless Packing, Microsoft, Microsoft Windows, MiniDisc, Modified discrete cosine transform, Monkey's Audio, MP3, MPEG Audio Decoder, MPEG Multichannel, MPEG-1 Audio Layer II, MPEG-4 SLS, Multi-Band Excitation, Musepack, Narrowband, Nellymoser, Nero Digital, Nippon Telegraph and Telephone, Ogg, Open-source model, OptimFROG, Opus (audio format), Original Sound Quality, Palm OS, Patent, Perian, PlayStation 3, PlayStation Portable, Polycom, POSIX, Proprietary software, Pulse-code modulation, Qualcomm code-excited linear prediction, QuickTime, QuteCom, RealAudio, RealPlayer, Reverse engineering, Rockbox, Sampling (signal processing), SBC (codec), Shorten (file format), SILK, Siren (codec), Skype, Skype for Business, Skype Technologies, Sony, Sony Dynamic Digital Sound, Sound Forge, Sound recording and reproduction, Speech coding, Speex, Stereophonic sound, Studio/transmitter link, Sub-band coding, Surround sound, SVOPC, Symbian, Telephony, TooLAME, Truespeech, TTA (codec), TwinVQ, Unix, Vaio, Variable bitrate, Variable-Rate Multimode Wideband, Vector sum excited linear prediction, VisualOn, Voice over IP, Vorbis, Walkman, WavPack, WebRTC, Wideband, Winamp, Windows legacy audio components, Windows Media Audio, Windows Media Components for QuickTime, Windows Media Encoder, Windows Media Player, Wireless microphone, Xing Technology, Xiph.Org Foundation, 3GPP, 8SVX. Expand index (149 more) » « Shrink index
An A-law algorithm is a standard companding algorithm, used in European 8-bit PCM digital communications systems to optimize, i.e. modify, the dynamic range of an analog signal for digitizing.
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.
Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required data bandwidth for a given signal-to-noise ratio.
The Adaptive Multi-Rate (AMR or AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding.
Adaptive Multi-Rate Wideband (AMR-WB) is a patented wideband speech audio coding standard developed based on Adaptive Multi-Rate encoding, using similar methodology as algebraic code excited linear prediction (ACELP).
Adaptive Transform Acoustic Coding (ATRAC) is a family of proprietary audio compression algorithms developed by Sony.
Adobe Flash is a deprecated multimedia software platform used for production of animations, rich Internet applications, desktop applications, mobile applications, mobile games and embedded web browser video players.
Adobe Flash Player (labeled Shockwave Flash in Internet Explorer and Firefox) is freeware for using content created on the Adobe Flash platform, including viewing multimedia contents, executing rich Internet applications, and streaming audio and video.
Advanced Audio Coding (AAC) is a proprietary audio coding standard for lossy digital audio compression.
Algebraic code-excited linear prediction (ACELP) is a patented speech coding algorithm by VoiceAge Corporation in which a limited set of pulses is distributed as excitation to a linear prediction filter.
In mathematics and computer science, an algorithm is an unambiguous specification of how to solve a class of problems.
aMSN is a free Windows Live Messenger clone.
Android is a mobile operating system developed by Google, based on a modified version of the Linux kernel and other open source software and designed primarily for touchscreen mobile devices such as smartphones and tablets.
Apple Inc. is an American multinational technology company headquartered in Cupertino, California, that designs, develops, and sells consumer electronics, computer software, and online services.
Apple Lossless, also known as Apple Lossless Audio Codec (ALAC), or Apple Lossless Encoder (ALE), is an audio coding format, and its reference audio codec implementation, developed by Apple Inc. for lossless data compression of digital music.
In digital audio data reduction technology, aptX (formerly apt-X) is a family of proprietary audio codec compression algorithms currently owned by Qualcomm.
Asao (also known as Nellymoser audio codec) is a proprietary single-channel (mono) codec and compression format optimized for low-bitrate transmission of audio, developed by Nellymoser Inc.
Asterisk is a software implementation of a telephone private branch exchange (PBX).
Advanced Television Systems Committee (ATSC) standards are a set of standards for digital television transmission over terrestrial, cable, and satellite networks.
An audio codec is a codec (a device or computer program capable of encoding or decoding a digital data stream) that encodes or decodes audio.
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).
MPEG-4 Audio Lossless Coding, also known as MPEG-4 ALS, is an extension to the MPEG-4 Part 3 audio standard to allow lossless audio compression.
Berkeley Software Distribution (BSD) was a Unix operating system derivative developed and distributed by the Computer Systems Research Group (CSRG) of the University of California, Berkeley, from 1977 to 1995.
In telecommunications and computing, bit rate (bitrate or as a variable R) is the number of bits that are conveyed or processed per unit of time.
Bluetooth is a wireless technology standard for exchanging data over short distances (using short-wavelength UHF radio waves in the ISM band from 2.4 to 2.485GHz) from fixed and mobile devices, and building personal area networks (PANs).
The Bluetooth Special Interest Group (Bluetooth SIG) is the standards organisation that oversees the development of Bluetooth standards and the licensing of the Bluetooth technologies and trademarks to manufacturers.
Broadcom Inc. (formerly Avago Technologies) is a designer, developer and global supplier of products based on analog and digital semiconductor technologies within four primary markets: wired infrastructure, wireless communications, enterprise storage, and industrial & others.
Founded in 1986, La Communauté électroacoustique canadienne / The Canadian Electroacoustic Community (CEC) is Canada’s national electroacoustic / computer music / sonic arts organization and as such is dedicated to promoting this progressive art form in its broadest definition: from “pure” acousmatic and computer music to soundscape and sonic art to hardware hacking and beyond.
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.
Chromium OS is an open-source operating system designed for running web applications and browsing the World Wide Web.
Cisco Systems, Inc. is an American multinational technology conglomerate headquartered in San Jose, California, in the center of Silicon Valley, that develops, manufactures and sells networking hardware, telecommunications equipment and other high-technology services and products.
The Sony CLIÉ is a series of personal digital assistants running the Palm Operating System developed and marketed by Sony from 2000 to 2005.
Code-excited linear prediction (CELP) is a speech coding algorithm originally proposed by M. R. Schroeder and B. S. Atal in 1985.
Codec 2 is a low-bitrate speech audio codec (speech coding) that is patent free and open source.
A codec listening test is a scientific study designed to compare two or more lossy audio codecs, usually with respect to perceived fidelity or compression efficiency.
In telecommunication and signal processing companding (occasionally called compansion) is a method of mitigating the detrimental effects of a channel with limited dynamic range.
The following comparison of audio players compares general and technical information for a number of software media player programs.
Α video codec is software or a device that provides encoding and decoding for digital video, and which may or may not include the use of video compression and/or decompression.
This table compares features of container formats (video file formats).
The following comparison of video players compares general and technical information for notable software media player programs.
Constant bitrate (CBR) is a term used in telecommunications, relating to the quality of service.
The cook codec is a lossy audio compression codec developed by RealNetworks.
Core Audio is a low-level API for dealing with sound in Apple's macOS and iOS operating systems.
Digital audio broadcasting (DAB) is a digital radio standard for broadcasting digital audio radio services, used in many countries across Europe, Asia, and the Pacific.
A container or wrapper format is a metafile format whose specification describes how different elements of data and metadata coexist in a computer file.
Digital rights management (DRM) is a set of access control technologies for restricting the use of proprietary hardware and copyrighted works.
Digital television (DTV) is the transmission of television signals, including the sound channel, using digital encoding, in contrast to the earlier television technology, analog television, in which the video and audio are carried by analog signals.
Digital Video Broadcasting (DVB) is a set of internationally open standards for digital television.
tags in this are generally true, just difficult to source due to the technical nature; don't remove unless 100% sure.
Dolby Atmos is the name of a surround sound technology announced by Dolby Laboratories in April 2012 and released in June that year, first utilized in Disney and Pixar's animated film Brave.
Dolby Digital is the name for audio compression technologies developed by Dolby Laboratories.
Dolby Digital Plus, also known as Enhanced AC-3 (and commonly abbreviated as DD+ or E-AC-3, or EC-3) is a digital audio compression scheme developed by Dolby Labs for transport and storage of multi-channel digital audio.
Dolby Laboratories, Inc. (often shortened to Dolby Labs) is a British-American company specializing in audio noise reduction and audio encoding/compression.
Dolby TrueHD is a lossless multi-channel audio codec developed by Dolby Laboratories which is used in home-entertainment equipment such as Blu-ray Disc players and A/V receivers.
DTS (Dedicated To Sound) is a series of multichannel audio technologies owned by Xperi Corporation (formerly known as Digital Theater Systems, Inc.), an American company specializing in digital surround sound formats used for both commercial/theatrical and consumer grade applications.
DTS-HD Master Audio (DTS-HD MA) is a combined lossless/lossy audio codec created by DTS (formerly Digital Theater Systems), commonly used for surround-sound movie soundtracks on Blu-ray Disc.
DVD (an abbreviation of "digital video disc" or "digital versatile disc") is a digital optical disc storage format invented and developed by Philips and Sony in 1995.
A DVD player is a device that plays DVD discs produced under both the DVD-Video and DVD-Audio technical standards, two different and incompatible standards.
Ekiga (formerly called GnomeMeeting) is a VoIP and video conferencing application for GNOME and Microsoft Windows.
An encoder is a device, circuit, transducer, software program, algorithm or person that converts information from one format or code to another, for the purposes of standardization, speed or compression.
Enhanced Full Rate or EFR or GSM-EFR or GSM 06.60 is a speech coding standard that was developed in order to improve the quite poor quality of GSM-Full Rate (FR) codec.
The European Telecommunications Standards Institute (ETSI) is an independent, not-for-profit, standardization organization in the telecommunications industry (equipment makers and network operators) in Europe, headquartered in Sophia-Antipolis, France, with worldwide projection.
Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB.
FAAC or Freeware Advanced Audio Coder is a software project which includes the AAC encoder FAAC and decoder FAAD2.
FairPlay was a digital rights management (DRM) technology developed by Apple Inc. It is built into the MP4 multimedia file format as an encrypted AAC audio layer, and is used by the company to protect copyrighted works sold through iTunes Store, allowing only authorized devices to play the content.
ffdshow is a codec mainly used for decoding of video in the MPEG-4 ASP (e.g. encoded with DivX or Xvid) and H.264/MPEG-4 AVC video formats, but it supports numerous other video and audio formats as well.
FFmpeg is a free software project, the product of which is a vast software suite of libraries and programs for handling video, audio, and other multimedia files and streams.
FLAC (Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation.
Flip4Mac from Telestream, Inc.
Fraunhofer FDK AAC (Full title Fraunhofer FDK AAC Codec Library for Android) 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.
The Fraunhofer Society (Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V., "Fraunhofer Society for the Advancement of Applied Research") is a German research organization with 69institutes spread throughout Germany, each focusing on different fields of applied science (as opposed to the Max Planck Society, which works primarily on basic science).
FreeSWITCH is a free and open source application server for real-time communication, WebRTC, telecommunications, video and Voice over Internet Protocol (VoIP).
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system.
G.711 is an ITU-T standard for audio companding.
G.718 is an ITU-T recommendation embedded scalable speech and audio codec providing high quality narrowband (250 Hz to 3.5 kHz) speech over the lower bit rates and high quality wideband (50 Hz to 7 kHz) speech over the complete range of bit rates.
G.719 is an ITU-T standard audio coding format providing high quality, moderate bit rate (32 to 128 kbit/s) wideband (20 Hz - 20 kHz audio bandwidth, 48 kHz audio sample rate) audio coding at low computational load.
G.722 is an ITU-T standard 7 kHz Wideband audio codec operating at 48, 56 and 64 kbit/s.
G.722.1 is a licensed royalty-free ITU-T standard audio codec providing high quality, moderate bit rate (24 and 32 kbit/s) wideband (50 Hz – 7 kHz audio bandwidth, 16 ksps (kilo-samples per second) audio coding. It is a partial implementation of Siren 7 audio coding format (which offers bit rates 16, 24, 32 kbit/s) developed by PictureTel Corp. (now Polycom, Inc.). Its official name is Low-complexity coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss. G.722.1 Annex C (or G.722.1C) is a low-complexity extension mode to G.722.1, which doubles the algorithm to permit 14 kHz audio bandwidth using a 32 kHz audio sample rate, at 24, 32, and 48 kbit/s. It is included in the official ITU-T Recommendation G.722.1. The name of this annex is Annex C – 14 kHz mode at 24, 32, and 48 kbit/s. It is an implementation of the mono version of Polycom's Siren 14 audio coding format. G.722.1 is the successor to PT716plus developed by PictureTel Corp. (now Polycom, Inc.), which has been used in videoconferencing systems for many years. As ITU-T Recommendation G.722.1, it was approved on September 30, 1999 after a four-year selection process involving extensive testing. G.722.1/Annex C was approved by ITU-T on May 14, 2005. G.722.1 is a transform-based compressor that is optimized for both speech and music. The G.722.1 algorithm is based on lapped transform technology, using a Modulated Lapped Transform (MLT). The computational complexity is quite low (5.5 floating-point MIPS) for an efficient high-quality compressor, and the algorithmic delay end-to-end is 40 ms. The numbering of the wideband ITU audio codecs is sometimes confusing. There are three principal codecs, which are unrelated, but all carrying the G.722 label. G.722 is the original 7 kHz codec, using ADPCM and operating at 48–64 kbit/s. G.722.1, another 7 kHz codec, operates at half the data rate while delivering comparable or better quality than G.722, but is a transform-based codec. G.722.1 Annex C is very similar to G.722.1, but provides twice the audio bandwidth, 14 kHz. And G.722.2, which operates on wideband speech and delivers very low bitrates, is an ACELP-based algorithm.
G.723 is an ITU-T standard speech codec using extensions of G.721 providing voice quality covering 300 Hz to 3400 Hz using Adaptive Differential Pulse Code Modulation (ADPCM) to 24 and 40 kbit/s for digital circuit multiplication equipment (DCME) applications.
G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames.
G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s.
G.728 is an ITU-T standard for speech coding operating at 16 kbit/s.
G.729 is a royalty-free narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds.
G.729.1 is an 8-32 kbit/s embedded speech and audio codec providing bitstream interoperability with G.729, G.729 Annex A and G.729 Annex B. Its official name is G.729-based embedded variable bit rate codec: An 8-32 kbit/s scalable wideband coder bitstream interoperable with G.729.
GStreamer is a pipeline-based multimedia framework that links together a wide variety of media processing systems to complete complex workflows.
Half Rate (HR or GSM-HR or GSM 06.20) is a speech coding system for GSM, developed in the early 1990s.
Harmonic Vector Excitation Coding, abbreviated as HVXC is a speech coding algorithm specified in MPEG-4 Part 3 (MPEG-4 Audio) standard for very low bit rate speech coding.
Harris Corporation is an American technology company, defense contractor and information technology services provider that produces wireless equipment, tactical radios, electronic systems, night vision equipment and both terrestrial and spaceborne antennas for use in the government, defense and commercial sectors.
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.
The International Electrotechnical Commission (IEC; in French: Commission électrotechnique internationale) is an international standards organization that prepares and publishes International Standards for all electrical, electronic and related technologies – collectively known as "electrotechnology".
The International Organization for Standardization (ISO) is an international standard-setting body composed of representatives from various national standards organizations.
The Internet Engineering Task Force (IETF) develops and promotes voluntary Internet standards, in particular the standards that comprise the Internet protocol suite (TCP/IP).
Internet Low Bitrate Codec (iLBC) is an open source royalty-free narrowband speech audio coding format codec and reference implementation, developed by Global IP Solutions (GIPS) formerly Global IP Sound (acquired by Google Inc in 2011).
internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS) (acquired by Google Inc in 2011).
iOS (formerly iPhone OS) is a mobile operating system created and developed by Apple Inc. exclusively for its hardware.
ISO base media file format (ISO/IEC 14496-12 – MPEG-4 Part 12) defines a general structure for time-based multimedia files such as video and audio.
The ITU Telecommunication Standardization Sector (ITU-T) is one of the three sectors (divisions or units) of the International Telecommunication Union (ITU); it coordinates standards for telecommunications.
iTunes is a media player, media library, Internet radio broadcaster, and mobile device management application developed by Apple Inc. It was announced on January 9, 2001.
Fraunhofer l3enc was the first public software able to encode PCM (.wav) files to the MP3 format.
LAME is a software encoder that converts audio to the MP3 file format.
Latency refers to a short period of delay (usually measured in milliseconds) between when an audio signal enters and when it emerges from a system.
Linear prediction is a mathematical operation where future values of a discrete-time signal are estimated as a linear function of previous samples.
Linear predictive coding (LPC) is a tool used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model.
Linux is a family of free and open-source software operating systems built around the Linux kernel.
The following is a list of compression formats and related codecs.
This is a listing of open-source implementations of media formats—usually called codecs.
A logarithmic scale is a nonlinear scale used when there is a large range of quantities.
In GSM, a Regular Pulse Excitation-Long Term Prediction (RPE-LTP) scheme is employed in order to reduce the amount of data sent between the mobile station (MS) and base transceiver station (BTS).
Lossless compression is a class of data compression algorithms that allows the original data to be perfectly reconstructed from the compressed data.
Lossless predictive audio compression (LPAC) is an improved lossless audio compression algorithm developed by Tilman Liebchen, Marcus Purat and Peter Noll at, Technical University Berlin (TU Berlin), to compress PCM audio in a lossless manner, unlike conventional audio compression algorithms which are lossy.
In information technology, lossy compression or irreversible compression is the class of data encoding methods that uses inexact approximations and partial data discarding to represent the content.
macOS (previously and later) is a series of graphical operating systems developed and marketed by Apple Inc. since 2001.
Media Foundation (MF) is a COM-based multimedia framework pipeline and infrastructure platform for digital media in Windows Vista, Windows 7, Windows 8, Windows 8.1 and Windows 10.
Meridian Lossless Packing, also known as Packed PCM (PPCM), is a lossless compression technique for compressing PCM audio data developed by Meridian Audio, Ltd..
Microsoft Corporation (abbreviated as MS) is an American multinational technology company with headquarters in Redmond, Washington.
Microsoft Windows is a group of several graphical operating system families, all of which are developed, marketed, and sold by Microsoft.
MiniDisc (MD) is a magneto-optical disc-based data storage format offering a capacity of 74 minutes and, later, 80 minutes, of digitized audio or 1 gigabyte of Hi-MD data.
The modified discrete cosine transform (MDCT) is a lapped 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.
Monkey's Audio is an algorithm and file format for lossless audio data compression.
MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is an audio coding format for digital audio.
MPEG Audio Decoder (MAD) is a GPL library for decoding files that have been encoded with an MPEG audio codec.
MPEG Multichannel is an extension to the MPEG-1 Layer II audio compression specification, as defined in the MPEG-2 Audio standard (ISO/IEC 13818-3) which allows it provide up to 5.1-channels (surround sound) of audio.
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM) 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).
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).
Multi-Band Excitation (MBE) is a series of proprietary speech coding standards developed by Digital Voice Systems, Inc.
Musepack or MPC is an open source lossy audio codec, specifically optimized for transparent compression of stereo audio at bitrates of 160–180 (manual set allows bitrates up to 320) kbit/s.
In radio, narrowband describes a channel in which the bandwidth of the message does not significantly exceed the channel's coherence bandwidth.
Nero Digital is a brand name applied to a suite of MPEG-4-compatible video and audio compression codecs developed by Nero AG of Germany and Ateme of France.
, commonly known as NTT, is a Japanese telecommunications company headquartered in Tokyo, Japan.
Ogg is a free, open container format maintained by the Xiph.Org Foundation.
The open-source model is a decentralized software-development model that encourages open collaboration.
OptimFROG is a proprietary lossless audio data compression codec developed by Florin Ghido.
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.
Original Sound Quality (OSQ) is an audio file format developed in 2002 by Steinberg Media Technologies GmbH and implemented e.g. in their audio editing software Wavelab 4 (and following releases) for lossless audio data compression.
Palm OS (also known as Garnet OS) is a discontinued mobile operating system initially developed by Palm, Inc., for personal digital assistants (PDAs) in 1996.
A patent is a set of exclusive rights granted by a sovereign state or intergovernmental organization to an inventor or assignee for a limited period of time in exchange for detailed public disclosure of an invention.
Perian is a discontinued open source QuickTime component that enabled Apple Inc.’s QuickTime to play several popular video formats not supported natively by QuickTime on macOS.
The PlayStation 3 (PS3) is a home video game console developed by Sony Computer Entertainment.
The PlayStation Portable (PSP) is a handheld game console developed by Sony Computer Entertainment.
Polycom is an American multinational corporation that develops video, voice and content collaboration and communication technology.
The Portable Operating System Interface (POSIX) is a family of standards specified by the IEEE Computer Society for maintaining compatibility between operating systems.
Proprietary software is non-free computer software for which the software's publisher or another person retains intellectual property rights—usually copyright of the source code, but sometimes patent rights.
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals.
Qualcomm code-excited linear prediction (QCELP), also known as Qualcomm PureVoice, is a speech codec developed in 1994 by Qualcomm to increase the speech quality of the IS-96A codec earlier used in CDMA networks.
QuickTime is an extensible multimedia framework developed by Apple Inc., capable of handling various formats of digital video, picture, sound, panoramic images, and interactivity.
QuteCom (previously called WengoPhone) is a free-software SIP-compliant VoIP client developed by the QuteCom (previously OpenWengo) community under the GNU General Public License (GPL).
RealAudio is a proprietary audio format developed by RealNetworks and first released in April 1995.
RealPlayer, formerly RealAudio Player, RealOne Player and RealPlayer G2, is a cross-platform media player app, developed by RealNetworks.
Reverse engineering, also called back engineering, is the process by which a man-made object is deconstructed to reveal its designs, architecture, or to extract knowledge from the object; similar to scientific research, the only difference being that scientific research is about a natural phenomenon.
Rockbox is a free and open-source software replacement for the OEM firmware in various forms of digital audio players (DAPs) with an original kernel.
In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal.
SBC, or low-complexity subband codec, is an audio subband codec specified by the Bluetooth Special Interest Group (SIG) for the Advanced Audio Distribution Profile (A2DP).
Shorten (SHN) is a file format used for compressing audio data.
SILK is an audio compression format and audio codec developed by Skype Limited.
Siren is a family of patented, transform-based, wideband audio coding formats and their audio codec implementations developed and licensed by PictureTel Corporation (acquired by Polycom, Inc. in 2001).
Skype is a telecommunications application software product that specializes in providing video chat and voice calls between computers, tablets, mobile devices, the Xbox One console, and smartwatches via the Internet and to regular telephones.
Skype for Business (formerly Microsoft Office Communicator and Microsoft Lync) is an instant messaging client used with Skype for Business Server or with Skype for Business Online (available with Microsoft Office 365).
Skype Technologies S.A.R.L (also known as Skype Software S.A.R.L, Skype Communications S.A.R.L, Skype Inc., and Skype Limited) is a telecommunications company headquartered in Luxembourg City, Luxembourg and Palo Alto, CA, United States, whose chief business is the manufacturing and marketing of the video chat and instant messaging computer software program Skype, and various Internet telephony services associated with it.
is a Japanese multinational conglomerate corporation headquartered in Kōnan, Minato, Tokyo.
is a cinema sound system developed by Sony, from which compressed digital sound information is recorded on both outer edges of the 35 mm film release print.
Sound Forge Audio Studio 12 (formerly known as Sonic Foundry Sound Forge, and later as Sony Sound Forge) is a digital audio editing suite by Magix Software GmbH which is aimed at the professional and semi-professional markets.
Sound recording and reproduction is an electrical, mechanical, electronic, or digital inscription and re-creation of sound waves, such as spoken voice, singing, instrumental music, or sound effects.
Speech coding is an application of data compression of digital audio signals containing speech.
Speex is an audio compression format specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on VoIP applications and podcasts.
Stereophonic sound or, more commonly, stereo, is a method of sound reproduction that creates an illusion of multi-directional audible perspective.
A studio/transmitter link (or STL) sends a radio station's or television station's audio and video from the broadcast studio or origination facility to a radio transmitter, television transmitter or uplink facility in another location.
In signal processing, sub-band coding (SBC) is any form of transform coding that breaks a signal into a number of different frequency bands, typically by using a fast Fourier transform, and encodes each one independently.
Surround sound is a technique for enriching the sound reproduction quality of an audio source with additional audio channels from speakers that surround the listener (surround channels).
SVOPC (Sinusoidal Voice Over Packet Coder) is a compression method for audio which is used by VOIP applications.
Symbian is a discontinued mobile operating system (OS) and computing platform designed for smartphones.
Telephony is the field of technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data, between distant parties.
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng.
Truespeech is a proprietary audio codec produced by the DSP Group.
True Audio (TTA) is a lossless compressor for multichannel 8, 16 and 24 bits audio data.
TwinVQ (transform-domain weighted interleave vector quantization) is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories (now Cyber Space Laboratories) in 1994.
Unix (trademarked as UNIX) is a family of multitasking, multiuser computer operating systems that derive from the original AT&T Unix, development starting in the 1970s at the Bell Labs research center by Ken Thompson, Dennis Ritchie, and others.
VAIO Corporation (standing for Visual Audio Intelligent Organizer), which is headquartered in Azumino, Nagano in Japan, is a manufacturer of personal computers.
Variable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding.
Variable-Rate Multimode Wideband (VMR-WB) is a source-controlled variable-rate multimode codec designed for robust encoding/decoding of wideband/narrowband speech.
Vector sum excited linear prediction (VSELP) is a speech coding method used in several cellular standards.
VisualOn is a Silicon Valley-based multimedia software company that provides high-definition audio and video entertainment to smartphones, tablets, laptops, connected TVs and other mobile and convergent devices.
Voice over Internet Protocol (also voice over IP, VoIP or IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.
Vorbis is a free and open-source software project headed by the Xiph.Org Foundation.
Walkman is a Sony brand tradename, originally used for portable audio cassette players from the late 1970s onwards.
WavPack is a free and open-source lossless audio compression format.
WebRTC (Web Real-Time Communication) is a free, open-source project that provides web browsers and mobile applications with real-time communication (RTC) via simple application programming interfaces (APIs).
In communications, a system is wideband when the message bandwidth significantly exceeds the coherence bandwidth of the channel.
Winamp is a media player for Windows, macOS and Android, originally developed by Justin Frankel and Dmitry Boldyrev by their company Nullsoft, which they later sold to AOL in 1999 for $80 million.
This article describes audio APIs and components in Microsoft Windows which are now obsolete or deprecated.
Windows Media Audio (WMA) is the name of a series of audio codecs and their corresponding audio coding formats developed by Microsoft.
Windows Media Components for QuickTime, also known as Flip4Mac WMV Player by Telestream, Inc. is one of the few commercial products that allow playback of Microsoft's proprietary audio and video codecs inside QuickTime for macOS.
Windows Media Encoder is a discontinued, freeware media encoder developed by Microsoft which enables content developers to convert or capture both live and prerecorded audio, video, and computer screen images to Windows Media formats for live and on-demand delivery.
Windows Media Player (WMP) is a media player and media library application developed by Microsoft that is used for playing audio, video and viewing images on personal computers running the Microsoft Windows operating system, as well as on Pocket PC and Windows Mobile-based devices.
A wireless microphone is a microphone without a physical cable connecting it directly to the sound recording or amplifying equipment with which it is associated.
Xing Technology was a live audio broadcast software company founded in Arroyo Grande, California in 1989 by former networking executive Howard Gordon.
Xiph.Org Foundation is a non-profit organization that produces free multimedia formats and software tools.
The 3rd Generation Partnership Project (3GPP) is a collaboration between groups of telecommunications standards associations, known as the Organizational Partners.
8-Bit Sampled Voice (8SVX) is an audio file format standard developed by Electronic Arts for the Commodore-Amiga computer series.