SettingsAudio codecs
A-law algorithm
An A-law algorithm is a standard companding algorithm, used in European digital communications systems to optimize, i.e., modify, the dynamic range of an analog signal for digitizing.
An A-law algorithm is a standard companding algorithm, used in European digital communications systems to optimize, i.e., modify, the dynamic range of an analog signal for digitizing.
AAC-LD
The MPEG-4 Low Delay Audio Coder (aka AAC Low Delay, or AAC-LD) is audio compression format designed to combine the advantages of perceptual audio coding with the low delay necessary...
The MPEG-4 Low Delay Audio Coder (aka AAC Low Delay, or AAC-LD) is audio compression format designed to combine the advantages of perceptual audio coding with the low delay necessary...
ACT (audio format)
ACT is a lossy ADPCM 8 kbit/s compressed audio format recorded by most Chinese MP3 and MP4 players with a recording function, and voice recorders.
ACT is a lossy ADPCM 8 kbit/s compressed audio format recorded by most Chinese MP3 and MP4 players with a recording function, and voice recorders.
Adaptive differential pulse-code modulation
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 t...
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 t...
Adaptive DPCM
Adaptive DPCM (ADPCM) is a variant of DPCM (differential pulse-code modulation) that varies the size of the quantization step, to allow further reduction of the required bandwidth for a gi...
Adaptive DPCM (ADPCM) is a variant of DPCM (differential pulse-code modulation) that varies the size of the quantization step, to allow further reduction of the required bandwidth for a gi...
Adaptive Multi-Rate Wideband
Adaptive Multi-Rate Wideband is a patented speech coding standard developed based on Adaptive Multi-Rate encoding, using similar methodology as Algebraic Code Excited Linear Prediction.
Adaptive Multi-Rate Wideband is a patented speech coding standard developed based on Adaptive Multi-Rate encoding, using similar methodology as Algebraic Code Excited Linear Prediction.
Adaptive Transform Acoustic Coding
Adaptive Transform Acoustic Coding is a family of proprietary audio compression algorithms developed by Sony.
Adaptive Transform Acoustic Coding is a family of proprietary audio compression algorithms developed by Sony.
Advanced Audio Coding
Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for digital audio.
Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for digital audio.
ADX (file format)
ADX is a lossy proprietary audio storage and compression format developed by CRI Middleware specifically for use in video games, it is derived from ADPCM. Its most notable feature is a looping f...
ADX is a lossy proprietary audio storage and compression format developed by CRI Middleware specifically for use in video games, it is derived from ADPCM. Its most notable feature is a looping f...
apt-X
The digital audio data reduction technology now known as apt-X is a family of proprietary audio codec compression algorithms developed by APT Licensing.
The digital audio data reduction technology now known as apt-X is a family of proprietary audio codec compression algorithms developed by APT Licensing.
Asao (codec)
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.
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.
Au file format
The Au file format is a simple audio file format introduced by Sun Microsystems.
The Au file format is a simple audio file format introduced by Sun Microsystems.
Audio codec
The term audio codec has two meanings depending on the context: In software, an "audio codec" is a computer program implementing an algorithm that compresses and decompresses digital audio data ...
The term audio codec has two meanings depending on the context: In software, an "audio codec" is a computer program implementing an algorithm that compresses and decompresses digital audio data ...
Audio Video Standard
Audio Video Standard is a compression standard for digital audio and video, and is competing with H.264/AAC to potentially replace MPEG-2.
Audio Video Standard is a compression standard for digital audio and video, and is competing with H.264/AAC to potentially replace MPEG-2.
CELT
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 au...
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 au...
Client Music Synthesis
The Client Music Synthesis (CMS) is a technology conceived to transfer very large music files via internet (or via any communication means having similar data transmission characteristics).
The Client Music Synthesis (CMS) is a technology conceived to transfer very large music files via internet (or via any communication means having similar data transmission characteristics).
Continuously variable slope delta modulation
Continuously variable slope delta modulation is a voice coding method.
Continuously variable slope delta modulation is a voice coding method.
Cook Codec
The cook codec is a lossy audio compression codec developed by RealNetworks.
The cook codec is a lossy audio compression codec developed by RealNetworks.
Dialnorm
Dialnorm is the meta-data parameter that controls playback gain within the Dolby Laboratories Dolby Digital (AC-3) audio compression system.
Dialnorm is the meta-data parameter that controls playback gain within the Dolby Laboratories Dolby Digital (AC-3) audio compression system.
Digital eXtreme Definition
Digital eXtreme Definition is an audio encoding scheme for professional use that was developed for editing high-resolution recordings because DSD, the audio standard used on Super Audio CD is no...
Digital eXtreme Definition is an audio encoding scheme for professional use that was developed for editing high-resolution recordings because DSD, the audio standard used on Super Audio CD is no...
Digital Sound Factory
Digital Sound Factory is a sound design company that creates sound libraries, known as SoundFont libraries, for playback on synthesizers and computers compatible with Steinberg Cubase, Cakewalk ...
Digital Sound Factory is a sound design company that creates sound libraries, known as SoundFont libraries, for playback on synthesizers and computers compatible with Steinberg Cubase, Cakewalk ...
Digital Speech Standard
Digital Speech Standard (DSS) is a proprietary compressed digital audio file format defined by the International Voice Association, a co-operative venture by Olympus, Philips and Grundig.
Digital Speech Standard (DSS) is a proprietary compressed digital audio file format defined by the International Voice Association, a co-operative venture by Olympus, Philips and Grundig.
Digital Theater System
DTS is a series of multichannel audio technologies owned by DTS, Inc. (, formerly known as Digital Theater Systems, Inc.), a company specializing in digital surround sound formats used for...
DTS is a series of multichannel audio technologies owned by DTS, Inc. (, formerly known as Digital Theater Systems, Inc.), a company specializing in digital surround sound formats used for...
Dolby Digital
Dolby Digital is the name for audio compression technologies developed by Dolby Laboratories.
Dolby Digital is the name for audio compression technologies developed by Dolby Laboratories.
Dolby Digital Plus
Dolby Digital Plus (DD+ or E-AC-3 (Enhanced AC-3), and sometimes incorrectly as EC-3) is a digital audio compression scheme.
Dolby Digital Plus (DD+ or E-AC-3 (Enhanced AC-3), and sometimes incorrectly as EC-3) is a digital audio compression scheme.
Dolby Headphone
Dolby Headphone is a technology developed by Dolby Laboratories, sometimes referred to as Mobile Surround, which creates a virtual surround sound environment in real-time using any set of two ch...
Dolby Headphone is a technology developed by Dolby Laboratories, sometimes referred to as Mobile Surround, which creates a virtual surround sound environment in real-time using any set of two ch...
Dolby Laboratories
Dolby Laboratories, Inc., often shortened to Dolby Labs, is an American company specializing in audio noise reduction and audio encoding/compression.
Dolby Laboratories, Inc., often shortened to Dolby Labs, is an American company specializing in audio noise reduction and audio encoding/compression.
DSM CC
Digital storage media command and control is a toolkit for developing control channels associated with MPEG-1 and MPEG-2 streams.
Digital storage media command and control is a toolkit for developing control channels associated with MPEG-1 and MPEG-2 streams.
DTS (sound system)
DTS is a series of multichannel audio technologies owned by DTS, Inc. (, formerly known as Digital Theater Systems, Inc.), an American company specializing in digital sur...
DTS is a series of multichannel audio technologies owned by DTS, Inc. (, formerly known as Digital Theater Systems, Inc.), an American company specializing in digital sur...
Dynamic Resolution Adaptation
Dynamic Resolution Adaptation is an audio encoding specification developed by DigiRise Technology.
Dynamic Resolution Adaptation is an audio encoding specification developed by DigiRise Technology.
Elecard
Elecard is a technology company that provides software products for video and audio encoding, decoding, processing, receiving and transmission.
Elecard is a technology company that provides software products for video and audio encoding, decoding, processing, receiving and transmission.
Enhanced Audio Codec
Enhanced Audio Codec is an audio codec developed and owned by Beijing E-World, that uses a unique perceptual model, spectral band replication, to compress the audio signal by utilizing the redun...
Enhanced Audio Codec is an audio codec developed and owned by Beijing E-World, that uses a unique perceptual model, spectral band replication, to compress the audio signal by utilizing the redun...
Enhanced full rate
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 codec.
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 codec.
Ensonido
Ensonido is a real-time post processing algorithm that allows users to play back MP3 Surround files in standard headphones.
Ensonido is a real-time post processing algorithm that allows users to play back MP3 Surround files in standard headphones.
Extended Adaptive Multi-Rate - Wideband
Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB. It adds support for stereo signals and higher sampling rates.
Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB. It adds support for stereo signals and higher sampling rates.
Extended Adaptive Multi-Rate – Wideband
Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB. It adds support for stereo signals and higher sampling rates.
Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB. It adds support for stereo signals and higher sampling rates.
FAAC
FAAC or Freeware Advanced Audio Coder is a software project which includes the AAC encoder FAAC and decoder FAAD2.
FAAC or Freeware Advanced Audio Coder is a software project which includes the AAC encoder FAAC and decoder FAAD2.
Full Rate
Full Rate or FR or GSM-FR or GSM 06.10 was the first digital speech coding standard used in the GSM digital mobile phone system.
Full Rate or FR or GSM-FR or GSM 06.10 was the first digital speech coding standard used in the GSM digital mobile phone system.
G.711
G.711 is an ITU-T standard for audio companding.
G.711 is an ITU-T standard for audio companding.
G.718
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 t...
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 t...
G.719
G.719 is an ITU-T standard audio codec 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 comput...
G.719 is an ITU-T standard audio codec 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 comput...
G.722
G.722 is a ITU-T standard 7 kHz wideband speech codec operating at 48, 56 and 64 kbit/s.
G.722 is a ITU-T standard 7 kHz wideband speech codec operating at 48, 56 and 64 kbit/s.
G.722.1
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 seco...
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 seco...
G.723
G.723 is a 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 ...
G.723 is a 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 ...
G.723.1
G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames.
G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames.
G.726
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.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
G.728 is an ITU-T standard for speech coding operating at 16 kbit/s.
G.728 is an ITU-T standard for speech coding operating at 16 kbit/s.
G.729
G.729 is an audio data compression algorithm for voice that compresses digital voice in packets of 10 milliseconds duration.
G.729 is an audio data compression algorithm for voice that compresses digital voice in packets of 10 milliseconds duration.
G.729.1
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 b...
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 b...
Half Rate
Half Rate (HR or GSM-HR or GSM 06.20) is a speech coding system for GSM, developed in the early 1990s.
Half Rate (HR or GSM-HR or GSM 06.20) is a speech coding system for GSM, developed in the early 1990s.
Harmonic and Individual Lines and Noise
Harmonic and Individual Lines and Noise (HILN) is a parametric codec for audio.
Harmonic and Individual Lines and Noise (HILN) is a parametric codec for audio.
High-Definition Coding
HDC (Hybrid Digital Coding) with SBR (spectral band replication) is a proprietary lossy audio compression codec developed by iBiquity for use with HD Radio.
HDC (Hybrid Digital Coding) with SBR (spectral band replication) is a proprietary lossy audio compression codec developed by iBiquity for use with HD Radio.
High-Efficiency Advanced Audio Coding
High-Efficiency Advanced Audio Coding (HE-AAC) is a lossy data compression scheme for digital audio defined as a MPEG-4 Audio profile in ISO/IEC 14496-3.
High-Efficiency Advanced Audio Coding (HE-AAC) is a lossy data compression scheme for digital audio defined as a MPEG-4 Audio profile in ISO/IEC 14496-3.
Iklax
iKlax is a multitrack and interactive audio format which was developed in France by iKlax Media and the LaBRI through a project started in 2006.
iKlax is a multitrack and interactive audio format which was developed in France by iKlax Media and the LaBRI through a project started in 2006.
Internet Low Bit Rate Codec
Internet Low Bitrate Codec is a royalty-free narrowband speech codec, developed by Global IP Solutions formerly Global IP Sound.
Internet Low Bitrate Codec is a royalty-free narrowband speech codec, developed by Global IP Solutions formerly Global IP Sound.
internet Speech Audio Codec
internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS).
internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS).
L3enc
Fraunhofer l3enc was the first public software able to encode PCM files to the MP3 format.
Fraunhofer l3enc was the first public software able to encode PCM files to the MP3 format.
Linear predictive coding
Linear predictive coding 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...
Linear predictive coding 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...
MainConcept
MainConcept GmbH is a software company developing video/audio codecs and also applications and plug-ins related to video/audio encoding.
MainConcept GmbH is a software company developing video/audio codecs and also applications and plug-ins related to video/audio encoding.
Mean opinion score
Mean opinion score is a test that has been used for decades in telephony networks to obtain the human user's view of the quality of the network.
Mean opinion score is a test that has been used for decades in telephony networks to obtain the human user's view of the quality of the network.
MO3
MO3 is a music file format developed by Ian Luck.
MO3 is a music file format developed by Ian Luck.
MP3
MPEG Audio Layer III, more commonly referred to as MP3, is a patented digital audio encoding format using a form of lossy data compression.
MPEG Audio Layer III, more commonly referred to as MP3, is a patented digital audio encoding format using a form of lossy data compression.
MP3 Surround
MP3 Surround is an extension of MP3 for multi-channel audio support including 5.1 surround sound.
MP3 Surround is an extension of MP3 for multi-channel audio support including 5.1 surround sound.
mp3PRO
mp3PRO is an audio compression algorithm that combines the MP3 audio format with spectral band replication compression methods.
mp3PRO is an audio compression algorithm that combines the MP3 audio format with spectral band replication compression methods.
MPEG Audio Decoder
MPEG Audio Decoder is a GPL library to decode files that have been encoded with an MPEG audio codec, written by Robert Leslie and produced by Underbit Technologies.
MPEG Audio Decoder is a GPL library to decode files that have been encoded with an MPEG audio codec, written by Robert Leslie and produced by Underbit Technologies.
MPEG Multichannel
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 (s...
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 (s...
MPEG Surround
MPEG Surround (ISO/IEC 23003-1 or MPEG-D Part 1), also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo au...
MPEG Surround (ISO/IEC 23003-1 or MPEG-D Part 1), also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo au...
MPEG-1 Audio Layer II
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.
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.
MPEG-2
MPEG-2 is a standard for "the generic coding of moving pictures and associated audio information".
MPEG-2 is a standard for "the generic coding of moving pictures and associated audio information".
MPEG-3
MPEG-3 is the designation for a group of audio and video coding standards agreed upon by the Moving Picture Experts Group (MPEG) designed to handle HDTV signals at 1080p in the range of 20 to 40...
MPEG-3 is the designation for a group of audio and video coding standards agreed upon by the Moving Picture Experts Group (MPEG) designed to handle HDTV signals at 1080p in the range of 20 to 40...
MPEG-4
MPEG-4 is a method of defining compression of audio and visual (AV) digital data.
MPEG-4 is a method of defining compression of audio and visual (AV) digital data.
MPEG-4 Part 3
MPEG-4 Part 3 or MPEG-4 Audio (formally ISO/IEC 14496-3) is the third part of the ISO/IEC MPEG-4 international standard developed by Moving Picture Experts Group.
MPEG-4 Part 3 or MPEG-4 Audio (formally ISO/IEC 14496-3) is the third part of the ISO/IEC MPEG-4 international standard developed by Moving Picture Experts Group.
MPEG-D
MPEG Surround (ISO/IEC 23003-1 or MPEG-D Part 1), also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo au...
MPEG Surround (ISO/IEC 23003-1 or MPEG-D Part 1), also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo au...
Musepack
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) ...
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) ...
Nellymoser Asao Codec
The Nellymoser Asao codec is a proprietary single-channel codec and compression format optimized for low-bitrate transmission of audio, developed by Nellymoser Inc.
The Nellymoser Asao codec is a proprietary single-channel codec and compression format optimized for low-bitrate transmission of audio, developed by Nellymoser Inc.
Nero Digital
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.
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.
NICAM
NICAM stands for Near Instantaneous Companded Audio Multiplex.
NICAM stands for Near Instantaneous Companded Audio Multiplex.
Opus (codec)
Opus (originally Harmony) is a low-delay wideband codec intended for applications such as VoIP that will eventually be royalty-free.
Opus (originally Harmony) is a low-delay wideband codec intended for applications such as VoIP that will eventually be royalty-free.
PEAQ
PEAQ is a standardized algorithm for objectively measuring perceived audio quality, developed in 1994-1998 by a joint venture of experts within Task Group 6Q of the International Telecommunicati...
PEAQ is a standardized algorithm for objectively measuring perceived audio quality, developed in 1994-1998 by a joint venture of experts within Task Group 6Q of the International Telecommunicati...
Perceptual audio coder
Perceptual Audio Coder (PAC) is an algorithm, like MPEG's MP3 standard, used to compress digital audio by removing extraneous information not perceived by most people.
Perceptual Audio Coder (PAC) is an algorithm, like MPEG's MP3 standard, used to compress digital audio by removing extraneous information not perceived by most people.
Pulse-code modulation
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals.
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals.
QDesign
QDesign Corporation was a leader in digital audio compression technologies.
QDesign Corporation was a leader in digital audio compression technologies.
QDX
QDX is an old lossy audio compression format, designed for portable multimedia players with small amount of memory.
QDX is an old lossy audio compression format, designed for portable multimedia players with small amount of memory.
RealAudio
RealAudio is a proprietary audio format developed by RealNetworks and first released in 1995.
RealAudio is a proprietary audio format developed by RealNetworks and first released in 1995.
RTAudio
RTAudio is a Microsoft produced adaptive wide-band speech codec.
RTAudio is a Microsoft produced adaptive wide-band speech codec.
Sb0
.sb0 is a form of high rate sound used in the newer Ubisoft games.
.sb0 is a form of high rate sound used in the newer Ubisoft games.
Siren (codec)
Siren is a family of patented, transform-based, wideband audio codecs developed and licensed by PictureTel Corporation (acquired by Polycom, Inc. in 2001).
Siren is a family of patented, transform-based, wideband audio codecs developed and licensed by PictureTel Corporation (acquired by Polycom, Inc. in 2001).
Siren Codec
Siren 7 (or Siren7 or only Siren) provides 7 kHz audio, bit rates 16, 24, 32 kbit/s and sampling frequency 16 kHz. Siren is derived from PictureTel's PT716plus algorithm.
Siren 7 (or Siren7 or only Siren) provides 7 kHz audio, bit rates 16, 24, 32 kbit/s and sampling frequency 16 kHz. Siren is derived from PictureTel's PT716plus algorithm.
Sony Dynamic Digital Sound
Sony Dynamic Digital Sound (SDDS) (Japanese: ソニーダイナミックデジタルサウンド) is a cinema sound system developed by Sony.
Sony Dynamic Digital Sound (SDDS) (Japanese: ソニーダイナミックデジタルサウンド) is a cinema sound system developed by Sony.
SoundFont
SoundFont is a brand name that collectively refers to a file format and associated technology designed to bridge the gap between recorded and synthesized audio, especially for the purposes of co...
SoundFont is a brand name that collectively refers to a file format and associated technology designed to bridge the gap between recorded and synthesized audio, especially for the purposes of co...
Spectral band replication
Spectral band replication (SBR) is a technology to enhance audio or speech codecs, especially at low bit rates and is based on harmonic redundancy in the frequency domain.
Spectral band replication (SBR) is a technology to enhance audio or speech codecs, especially at low bit rates and is based on harmonic redundancy in the frequency domain.
Speex
Speex is a patent-free audio compression format designed for speech and also a free software speech codec that may be used on VoIP applications and podcasts.
Speex is a patent-free audio compression format designed for speech and also a free software speech codec that may be used on VoIP applications and podcasts.
SRS Labs
SRS Labs, Inc., is a Santa Ana, California-based audio technology engineering company that specializes in audio enhancement solutions for wide variety of consumer electronic devices.
SRS Labs, Inc., is a Santa Ana, California-based audio technology engineering company that specializes in audio enhancement solutions for wide variety of consumer electronic devices.
SVOPC
SVOPC (ang. Sinusoidal Voice Over Packet Coder) is a compression method for audio which is used by VOIP applications.
SVOPC (ang. Sinusoidal Voice Over Packet Coder) is a compression method for audio which is used by VOIP applications.
TooLAME
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng.
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng.
Transparency (data compression)
In data compression or psychoacoustics, transparency is the ideal result of lossy data compression.
In data compression or psychoacoustics, transparency is the ideal result of lossy data compression.
Tremor (software)
Tremor by the Xiph.Org Foundation is a fixed-point version of the Vorbis decoder for those platforms without floating point operations.
Tremor by the Xiph.Org Foundation is a fixed-point version of the Vorbis decoder for those platforms without floating point operations.
Truespeech
Truespeech is a proprietary audio codec produced by the DSP Group.
Truespeech is a proprietary audio codec produced by the DSP Group.
TwinVQ
TwinVQ (transform-domain weighted interleave vector quantization) is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories (no...
TwinVQ (transform-domain weighted interleave vector quantization) is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories (no...
Ultra Low Delay Audio Coder
The Ultra Low Delay Audio Coder is a development of the Fraunhofer Institute for Digital Media Technology, which is headed by one of the fathers of MP3, Karlheinz Brandenburg, and of the Fraunho...
The Ultra Low Delay Audio Coder is a development of the Fraunhofer Institute for Digital Media Technology, which is headed by one of the fathers of MP3, Karlheinz Brandenburg, and of the Fraunho...
Unified Speech and Audio Coding
Unified Speech and Audio Coding 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.
Unified Speech and Audio Coding 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.
VivoActive
VivoActive is an audio/video format created by Vivo Software, acquired by RealNetworks in 1997.
VivoActive is an audio/video format created by Vivo Software, acquired by RealNetworks in 1997.
Vorbis
Vorbis is a free software / open source project headed by the Xiph.Org Foundation (formerly Xiphophorus company).
Vorbis is a free software / open source project headed by the Xiph.Org Foundation (formerly Xiphophorus company).
Winamp Alternative
Winamp Alternative is a codec package for Microsoft Windows created for playing SHOUTcast streams with any media player that supports DirectShow and Nullsoft Streaming Video with major browsers.
Winamp Alternative is a codec package for Microsoft Windows created for playing SHOUTcast streams with any media player that supports DirectShow and Nullsoft Streaming Video with major browsers.
Windows Media Audio
Windows Media Audio (WMA) is an audio data compression technology developed by Microsoft.
Windows Media Audio (WMA) is an audio data compression technology developed by Microsoft.
Xiph QuickTime Components
The Xiph QuickTime Components are Xiph.org's implementation of the Ogg container along with the Speex, Theora, FLAC and Vorbis codecs for QuickTime.
The Xiph QuickTime Components are Xiph.org's implementation of the Ogg container along with the Speex, Theora, FLAC and Vorbis codecs for QuickTime.
XMA (audio format)
XMA is the native Xbox 360 compressed audio format, based on the WMA Pro architecture.
XMA is the native Xbox 360 compressed audio format, based on the WMA Pro architecture.
Μ-law algorithm
The μ-law algorithm is also used in the .au format, which dates back at least to the SPARCstation 1 as the native method used by Sun's /dev/audio interface, widely used as a de facto standard fo...
The μ-law algorithm is also used in the .au format, which dates back at least to the SPARCstation 1 as the native method used by Sun's /dev/audio interface, widely used as a de facto standard fo...
μ-law algorithm
The μ-law algorithm is also used in the .au format, which dates back at least to the SPARCstation 1 as the native method used by Sun's /dev/audio interface, widely used as a de facto standard fo...
The μ-law algorithm is also used in the .au format, which dates back at least to the SPARCstation 1 as the native method used by Sun's /dev/audio interface, widely used as a de facto standard fo...