Audio codecs

MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is an audio coding format developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg. It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners; for example, compared to CD-quality digital audio, MP3 compression can commonly achieve a 75–95% reduction in size, depending on the bit rate. In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3

Audio compression technologies

digital representation of sampled analog signals

MPEG-4 is a group of international standards for the compression of digital audio and visual data, multimedia systems, and file storage formats. It was originally introduced in late 1998 as a group of audio and video coding formats and related technology agreed upon by the ISO/IEC Moving Picture Experts Group (MPEG) (ISO/IEC JTC 1/SC29/WG11) under the formal standard ISO/IEC 14496 – Coding of audio-visual objects. Uses of MPEG-4 include compression of audiovisual data for Internet video and CD distribution, voice (telephone, videophone) and broadcast television applications. The MPEG-4 st

thumb|right|MPEG-2 is used in Digital Video Broadcast and DVDs. The MPEG transport stream, TS, and [[MPEG program stream, PS, are container formats.]]

Audio compression format from MPEG

Dolby Laboratories, Inc. (Dolby Labs or simply Dolby) is an American technology corporation specializing in audio noise reduction, audio encoding/compression, spatial audio, and high-dynamic-range television (HDR) imaging. Dolby licenses its technologies to consumer electronics manufacturers.

MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s (26:1 and 6:1 compression ratios respectively) without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical.

series of multichannel audio technologies owned by DTS, Inc.

audio data compression technology

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 a CD while storing audio information with minimal perceptible loss in quality. Improvements to the codec in the form of ATRAC3, ATRAC3plus, and ATRAC Advanced Lossless followed in 1999, 2002, and 2006 respectively.

Near Instantaneous Companded Audio Multiplex (NICAM) is an early form of lossy compression for digital audio. It was originally developed in the early 1970s for point-to-point links within broadcasting networks. In the 1980s, broadcasters began to use NICAM compression for transmissions of stereo TV sound to the public.

RealAudio, also spelled Real Audio, is a proprietary audio format developed by RealNetworks and first released in April 1995. It uses a variety of audio codecs, ranging from low-bitrate formats that could be used over dialup modems, to high-fidelity formats for music. It can be used as a streaming audio format, that is played at the same time as it is downloaded.

cinema sound system

The Speex project is an attempt to create a free software speech codec, unencumbered by patent restrictions. Speex is licensed under the BSD License and is used with the Xiph.org Foundation's Ogg container format.

audio coding format

technique used to encode voices in telephony

G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. It is an ITU-T standard (Recommendation) for audio encoding, titled Pulse code modulation (PCM) of voice frequencies released for use in 1972.

G.729 is a royalty-free narrow-band vocoder-based audio data compression algorithm using a frame length of . It is officially described as Coding of speech at 8 kbit/s using code-excited linear prediction speech coding (CS-ACELP), and was introduced in 1996. The wide-band extension of G.729 is called G.729.1, which equals G.729 Annex J.

speech analysis and encoding technique

system for digitally recreating audible signals

audio companding algorithm

file format

MPEG-3 was the designation for an abandoned plan to create 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 megabits per second. MPEG-3 was launched as an effort to address the need of an HDTV standard while work on MPEG-2 was underway, but it was soon discovered that MPEG-2, at high data rates, would accommodate HDTV. Thus, in 1992 HDTV was included as a separate profile in the MPEG-2 standard and MPEG-3 was rolled into MPEG-2.

mp3PRO is an unmaintained proprietary audio compression codec that combines the MP3 audio format with the spectral band replication (SBR) compression method. At the time it was developed it could reduce the size of a stereo MP3 by as much as 50% while maintaining the same relative quality. This works, fundamentally, by discarding the higher half of the frequency range and algorithmically replicating that information while decoding.

device or program that encodes/decodes audio data in some bitstream format

[[File:Soundfont Comparison (Updated).wav|thumb|Playing a single MIDI file while switching between several SoundFont files available on the Internet.SoundFont files used in the chronological order: • SONiVOX EAS GM Wavetable (Legacy Android Soundset)* [1 MB]• RLNDGM.sf2 (Microsoft GS Wavetable Synth)* [3 MB]• FluidR3 GM.sf2 [141 MB]• SGM-V2.01.sf2 [235 MB]• Orpheus_1.047.sf2* [1.18 GB]• ChoriumRevA.sf2 (Modified) [56 MB]• ColomboGMGS2 SoundFont v14.5 [245 MB] *Marked soundfonts fall back to play "Muted Guitar" at Bank 0. whereas the MIDI file addresses "Muted Distortion Guitar" at Bank 1 (SC-

algorithm

audio file format

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. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new 16 kbit/s rate. The four bit rates associated with G.726 are often referred to by the bit size of a sample, which are 2, 3, 4, and 5-bits respectively. The corresponding wide-band codec based on the same technology is G.722.

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. The compression technique has been used in both standardized and proprietary designs.

G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on the same technology is G.726.

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. The standard G.723 is obsolete and has been superseded by G.726.

SILK is an audio compression format and audio codec developed by Skype Limited, now a Microsoft subsidiary. It was developed for use in Skype, as a replacement for the SVOPC codec. Since licensing out, it has also been used by others. It has been extended to the Internet standard Opus codec.

speech coding standard

aptX (apt stands for audio processing technology) is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications.

video codec

third part of the ISO/IEC MPEG-4 international standard

low bitrate digital audio enhancement technique

G.728 is an ITU-T standard for speech coding operating at 16 kbit/s. It is officially described as Coding of speech at 16 kbit/s using low-delay code excited linear prediction.

speech coding standard

high-definition digital audio format

file format family

G.723.1 is an audio codec for voice that compresses voice audio in frames. An algorithmic look-ahead of duration means that total algorithmic delay is . Its official name is Dual rate speech coder for multimedia communications transmitting at 5.3 and . It is sometimes associated with a Truespeech trademark in coprocessors produced by DSP Group.

audio coded

software suite of audio/video codecs

audio data compression scheme optimized for speech coding

family of patented, transform-based, wideband audio coding formats and their audio codec implementations

G.729.1 is an 8- 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- scalable wideband coder bitstream interoperable with G.729. It was introduced in 2006.

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. It was produced through a collaboration between Polycom and Ericsson.

audio coding technology developed by Sony, which allows streaming audio over Bluetooth

speech coding standard

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. It uses a modified discrete cosine transform (MDCT) audio data compression algorithm.

audiophile lossy audio codec