Glossary of digital audio

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A~C

AAC
Advanced Audio Coding is a standardized, lossy compression and encoding scheme for digital audio. Designed to be the successor of the MP3 format, AAC generally achieves better sound quality than MP3 at similar bit rates.
AC-3
Audio Coding 3 is a 6-channel, audio file format by Dolby Laboratories that usually accompanies DVD viewing. It operates 5 channels for normal range speakers (20 to 20,000 Hz) and the 6th channel reserved for low-frequency (20 to 120 Hz) sub-woofer operation. AC3 increases fidelity over its previous surround sound standard, Pro-logic, with independent tracks for each of the 6 speakers, a 16-bit depth at 48 kHz sampling rate with a maximum bit rate of 640 kbit/s.
AIFF
Audio Interchange File Format is an audio file format standard used for storing sound data for personal computers and other electronic audio devices. The format was developed by Apple Computer.
ALAC
Apple Lossless Audio Codec is an audio codec developed by Apple Inc. for lossless data compression of digital music.
Audio bit depth
Bit depth or sample resolution is the number of bits of information recorded for each sample. Bit depth directly corresponds to the resolution of each sample in a set of digital audio data. Common examples of bit depth include CD quality audio, which is recorded at 16 bits, SACD with effective 20-bit resolution, and DVD-Audio, which can support up to 24-bit audio.
Bit rate
Represents the amount of information, or detail, that is stored per unit of time of a recording. Common examples of bit rates include MP3 which is recorded at 128–320 kbits/s, CD quality audio (LPCM) which is recorded at 1,411.2 kbit/s, SACD (DSD) which is recorded at 5,644.8 kbit/s, and DVD-Audio (MLP), which is recorded at 18,000 kbit/s.
CD
Compact Disc is a 16-bit, 44.1 kHz system, giving a frequency response of 5 to 22,000 Hz and 96 dB dynamic range.

D

DFF
The format used for SACD mastering and 5.1-channel downloads  file extension for DSDIFF files
DSD
Direct Stream Digital is a 1-bit method of storing audio on digital media. The audio format used for Super Audio CDs (SACD) with effective 20-bit resolution.
DSDIFF
Direct Stream Digital Interchange File Format  Format for the storage or exchange of one-bit delta sigma modulated audio, often called Direct Stream Digital, or for the losslessly compressed version called Direct Stream Transfer (DST).
DSF
Direct Stream File is a stereo-only, simplified form of DFF. a 1-bit 2-channel only audio file format developed and used by Sony for their VAIO personal and laptop computers equipped with Sound Reality Sigma-Delta DAC. A 1-bit professional audio file format used for SACD production.
DST
Direct Stream Transfer is lossless compression; part of MPEG-4; used for SACD
DTS
A series of multichannel audio technologies owned by DTS, Inc. (formerly known as Digital Theater Systems, Inc.). DTS supports bit rates up to 1,534 kbit/s, sampling rates up to 48.0 kHz and bit depths up to 24 bits.
DVD-A
DVD-Audio is a digital format for delivering high-fidelity audio content on a DVD. DVD-A is capable of frequencies from 0 to 96 kHz with a maximum dynamic range of 144 dB.
Dynamic range
The ratio of the amplitude of the loudest possible undistorted sine wave to the root mean square (rms) noise amplitude. The 16-bit compact disc has a theoretical dynamic range of about 96 dB (or about 98 dB for sinusoidal signals, per the formula). Digital audio with 20-bit digitization is theoretically capable of 120 dB dynamic range; similarly, 24-bit digital audio calculates to 144 dB dynamic range.

F–Z

FLAC
Free Lossless Audio Codec is a codec (compressor-decompressor) which allows digital audio to be losslessly compressed such that file size is reduced without any information being lost.
LPCM
Linear PCM is Pulse-code modulation (PCM) with linear quantization.
MKV
Matroska Multimedia Container is an open standard free container format, a file format that can hold an unlimited number of video, audio, picture, or subtitle tracks in one file.
MLP
Meridian Lossless Packing, also known as Packed PCM (PPCM), is a proprietary lossless compression technique for compressing PCM audio data developed by Meridian Audio, Ltd. MLP is the standard lossless compression method for DVD-Audio content (often advertised with the Advanced Resolution logo) and typically provides about 1.5:1 compression on most music material. All DVD-Audio players are equipped with MLP decoding, while its use on the discs themselves is at their producers' discretion. To store 5.1 tracks in 88.2 kHz / 20-bit, 88.2 kHz / 24-bit, 96 kHz / 20-bit or 96 kHz / 24-bit on a DVD disc, the use of MLP compression is mandatory.
[ clarification needed ]
MP3
MPEG-1 or MPEG-2 Audio Layer III is a patented encoding format for digital audio which uses a form of lossy data compression.
PCM
Pulse-code modulation is a method used to digitally represent sampled analog signals. It is the standard method of storing audio in computers and various Blu-ray, DVD and Compact Disc formats. PCM supports bit rates up to 1,534 kbit/s, sampling rates up to 48.0 kHz, and bit depths up to 24 bits.
Sampling rate
Sampling rate or sampling frequency is the number of samples per unit of time (usually seconds) taken from a continuous signal to make a discrete signal. Common examples of sampling rates include CD quality audio, which is recorded at 44.1 kHz, SACD which is recorded at 2.8224 MHz and DVD-Audio, which can support 96 kHz and higher.
SACD
Super Audio CD is a format capable of delivering a dynamic range of 120 dB from 20 to 20 kHz and an extended frequency response up to 100 kHz, although most currently available players list an upper limit of 80–90 kHz and 20 kHz is the upper limit of human hearing.
WAV
Waveform Audio File Format is a Microsoft and IBM audio file format standard for storing an audio bitstream on PCs.

See also

  1. Audio codec
  2. Audio compression (data)
  3. Audio file format
  4. Digital audio

Related Research Articles

An audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data is called the audio coding format and can be uncompressed, or compressed to reduce the file size, often using lossy compression. The data can be a raw bitstream in an audio coding format, but it is usually embedded in a container format or an audio data format with defined storage layer.

A codec is a device or computer program that encodes or decodes a data stream or signal. Codec is a portmanteau of coder/decoder.

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.

<span class="mw-page-title-main">Super Audio CD</span> Read-only optical disc for high-fidelity audio storage

Super Audio CD (SACD) is an optical disc format for audio storage introduced in 1999. It was developed jointly by Sony and Philips Electronics and intended to be the successor to the Compact Disc (CD) format.

Dolby Digital, originally synonymous with Dolby AC-3, is the name for what has now become a family of audio compression technologies developed by Dolby Laboratories. Formerly named Dolby Stereo Digital until 1995, the audio compression is lossy, based on the modified discrete cosine transform (MDCT) algorithm. The first use of Dolby Digital was to provide digital sound in cinemas from 35 mm film prints; today, it is also used for applications such as TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles.

<span class="mw-page-title-main">G.711</span> ITU-T recommendation

G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. G.711 passes audio signals in the range of 300–3400 Hz and samples them at the rate of 8,000 samples per second, with the tolerance on that rate of 50 parts per million (ppm). Non-uniform (logarithmic) quantization with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. There are two slightly different versions: μ-law, which is used primarily in North America and Japan, and A-law, which is in use in most other countries outside North America.

<span class="mw-page-title-main">DVD-Audio</span> DVD format for storing high-fidelity audio

DVD-Audio is a digital format for delivering high-fidelity audio content on a DVD. DVD-Audio uses most of the storage on the disc for high-quality audio and is not intended to be a video delivery format.

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 encoders at the same bit rate.

In telecommunications and computing, bit rate is the number of bits that are conveyed or processed per unit of time.

<span class="mw-page-title-main">Sound quality</span> Assessment of the audio output from an electronic device

Sound quality is typically an assessment of the accuracy, fidelity, or intelligibility of audio output from an electronic device. Quality can be measured objectively, such as when tools are used to gauge the accuracy with which the device reproduces an original sound; or it can be measured subjectively, such as when human listeners respond to the sound or gauge its perceived similarity to another sound.

<span class="mw-page-title-main">Direct Stream Digital</span> System for digitally encoding audio signals

Direct Stream Digital (DSD) is a trademark used by Sony and Philips for their system for digitally encoding audio signals for the Super Audio CD (SACD).

WavPack is a free and open-source lossless audio compression format and application implementing the format. It is unique in the way that it supports hybrid audio compression alongside normal compression which is similar to how FLAC works. It also supports compressing a wide variety of lossless formats, including various variants of PCM and also DSD as used in SACDs, together with its support for surround audio.

<span class="mw-page-title-main">DTS (company)</span> Series of multichannel audio technologies

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.

Dolby Digital Plus, also known as Enhanced AC-3 is a digital audio compression scheme developed by Dolby Labs for transport and storage of multi-channel digital audio. It is a successor to Dolby Digital (AC-3), also developed by Dolby, and has a number of improvements including support for a wider range of data rates, increased channel count and multi-program support, and additional tools (algorithms) for representing compressed data and counteracting artifacts. While Dolby Digital (AC-3) supports up to five full-bandwidth audio channels at a maximum bitrate of 640 kbit/s, E-AC-3 supports up to 15 full-bandwidth audio channels at a maximum bitrate of 6.144 Mbit/s.

Dolby TrueHD is a lossless, multi-channel audio codec developed by Dolby Laboratories for home video, used principally in Blu-ray Disc and compatible hardware. Dolby TrueHD, along with Dolby Digital Plus (E-AC-3) and Dolby AC-4, is one of the intended successors to the Dolby Digital (AC-3) lossy surround format. Dolby TrueHD competes with DTS's DTS-HD Master Audio, another lossless surround sound codec.

<span class="mw-page-title-main">DTS-HD Master Audio</span> Lossless audio codec for home theater

DTS-HD Master Audio is a multi-channel, lossless audio codec developed by DTS as an extension of the lossy DTS Coherent Acoustics codec. Rather than being an entirely new coding mechanism, DTS-HD MA encodes an audio master in lossy DTS first, then stores a concurrent stream of supplementary data representing whatever the DTS encoder discarded. This gives DTS-HD MA a lossy "core" able to be played back by devices that cannot decode the more complex lossless audio. DTS-HD MA's primary application is audio storage and playback for Blu-ray Disc media; it competes in this respect with Dolby TrueHD, another lossless surround format. DTS-HD MA has enjoyed the greater share of this market since 2010, with the notable exception of the TrueHD-encoded Dolby Atmos spatial surround format, which is more popular than DTS's competing DTS:X.

aptX Family of proprietary audio codecs owned by Qualcomm

aptX is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications.

Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.