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Digital audio comprises audio signals stored in a digital format. Specifically, the term encompasses the following: Sound reproduction is the electrical or mechanical re-creation and/or amplification of sound, often as music. ...
A digital system is one that uses discrete numbers, especially binary numbers, or non-numeric symbols such as letters or icons, for input, processing, transmission, storage, or display, rather than a continuous spectrum of values (an analog system). ...
- Audio conversion:
- Analogue to digital conversion (ADC) - the sampling and quantization of an analogue audio signal.
- Digital to analogue conversion (DAC) - the conversion of digital audio to a line level signal for playback or distribution.
- Audio signal processing - processing the digital signal in some way, such as to apply equalisation, reverberation, or to perform sample rate conversion.
- Storage, retrieval, and transmission of digital information in an audio format such as CD, MP3, Ogg Vorbis, etc.
An analog-to-digital converter (abbreviated ADC, A/D or A to D) is an electronic circuit that converts continuous signals to discrete digital numbers. ...
In signal processing, sampling is the reduction of a continuous signal to a discrete signal. ...
In sound processing, quantization is the process of aligning a set of musical notes to conform to a grid. ...
For the Analog Science Fiction and Science Fact publication, see Astounding Magazine. ...
In electronics, a digital-to-analog converter (DAC or D-to-A) is a device for converting a digital (usually binary) code to an analogue signal (current, voltage or charges). ...
Line level is the strength of an audio signal used to transmit analog sound information between audio components such as CD and DVD players, TVs, amplifiers, and mixing consoles. ...
This article or section does not cite its references or sources. ...
It has been suggested that this article or section be merged with Equalization filter. ...
When sound is produced in a space, multiple reflections may build up and blend together, creating reverberation, or reverb. ...
Sample rate conversion is the process of converting a (usually digital) signal from one sampling rate to another, while changing the information carried by the signal as little as possible. ...
An audio format is a medium for storing sound and music. ...
To meet Wikipedias quality standards, this article may require cleanup. ...
MPEG-1 Audio Layer 3, more commonly referred to as MP3, is a popular digital audio encoding and lossy compression format, 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. ...
This page is about the audio compression codec. ...
The digital paradigm Digital technology has emerged because of its supreme usefulness to sound recording, manipulation, mass-production and distribution. The modern day distribution of music across the internet through on-line stores depends on digital recording, and digital compression algorithms. "Dematerialization" of the music software into computer files has significantly reduced costs of distribution. However, it has brought about the concomitant rise in music sharing through peer to peer networks, which is illegal in many countries. The Recording Industry Association of America and other organizations claim that music sharing is severely harming the profitability of their business. Audio compression is a form of data compression designed to reduce the size of audio data files. ...
Flowcharts are often used to graphically represent algorithms. ...
A peer-to-peer (or P2P) computer network is a network that relies primarily on the computing power and bandwidth of the participants in the network rather than concentrating it in a relatively low number of servers. ...
The Recording Industry Association of America (or RIAA) is the trade group that represents the recording industry in the United States. ...
From the Long-play gramophone record and compact cassette, the 78 rpm vinyl records and wax cylinders before them, analogue audio music storage and reproduction have been based on the same principles upon which human hearing are based. Sounds begin and end as mechanical energy wave forms in air, are captured in said wave form, and transformed into an electrical energy by a microphone transducer. Although its nature may change, its fundamental wave-like characteristics remain unchanged during its storage, transformation, duplication, amplification. Up until very recently, analogue audio has been susceptible to significant information loss, as noise and distortions tend to creep in at each stage. The Beatles Magical Mystery Tour (1967) as a 33 â
LP vinyl record A gramophone record (also phonograph record, or simply record) is an analogue sound recording medium consisting of a flat disc with an inscribed modulated spiral groove starting near the periphery and ending near the centre of the disc. ...
The Compact Cassette logo inspired that of the Compact Disc two decades later. ...
Revolutions per minute (abbreviated rpm, RPM, r/min, or min-1) is a unit of frequency, commonly used to measure rotational speed, in particular in the case of rotation around a fixed axis. ...
Vinyl products (such as these records) come in many colors. ...
The earliest method of recording and reproducing sound was on cylinder phonograph recordings. ...
On the other hand, the digital audio chain begins when sound is converted into electrical signals - ‘on/off’ pulses – rather than electro-mechanical signals. The advantage of digital audio is the ability to be copied or transmitted more conveniently, and with arguably lower loss. This ability to control signal losses is important in a professional studio environment, where signals could pass many times through cables, mixing desks and processing equipment before the recording is finally mixed down onto a two-track master for manufacturing. BBC Local Radio Mark III radio mixing desk In professional audio, a mixing console, mixing desk (Brit. ...
Overview of digital audio Sound inherently begins and ends as an analogue signal, and in order for the benefits of digital audio to be realised, the integrity of the signal during transmission must be preserved. The conversion process at both ends of the chain must also be of low loss in order to ensure sonic fidelity. Sound is a disturbance of mechanical energy that propagates through matter as a wave. ...
Generation loss refers to the loss of quality between subsequent copies of data. ...
An ideal audio system reproduces source audio perfectly. Digital audio attempts to approximate this by generating audio which sounds as much like the original as possible. Since typical human ears can only detect a range of sounds between approximately 20 Hz and 20,000 Hz, digital audio systems only need to be able to generate sounds within this range. Thus, since they lose audio frequency information outside of the human range, digital systems are lossy. A lossy data compression method is one where compressing a file and then decompressing it retrieves a file that may well be different to the original, but is close enough to be useful in some way. ...
Digital information is also lost in transfer through misreading, but can be "restored" by error correction and interpolation circuitry. Put another way, the information is only lost during the conversion from analogue to digital (and vice versa), and the amount of loss can be more predictable. The restoration of the original music waveforms by decompression during playback should exactly mirror the compression process. However, upper harmonics which have been discarded can never be restored, with complete accuracy or otherwise. Upon its re-conversion into analogue via the amplifier/loudspeaker, the scheme relies heavily on the human brain to “fill in the gaps” – that is to say supply the missing sound during playback. This capability has been well discussed especially with respect to the brain supplying the fundamental frequency of a tone. The generally accepted frequency response of human hearing is from 20 Hz - 20 kHz. According to Nyquist, the maximum bandwidth that can be represented by a digital signal is less than half the sample rate. This leads to a required sample rate of at least 40 kHz. In practise, a slightly higher sample rate is needed to allow for a practical anti-aliasing filter. But, the Shannon equation for reconstructing the original data fully requires infinite samples. Frequency response is the measure of any systems response to frequency, but is usually used in connection with electronic amplifiers and similar systems, particularly in relation to audio signals. ...
The Nyquist-Shannon sampling theorem is the fundamental theorem in the field of information theory, in particular telecommunications. ...
Bandwidth is a measure of frequency range and is typically measured in hertz. ...
The sampling frequency or sampling rate defines the number of samples per second taken from a continuous signal to make a discrete signal. ...
An anti-aliasing filter is commonly used in conjuction with digital signal processing and is a filter to restrict the bandwidth to approximately satisfy the Shannon-Nyquist sampling theorem. ...
In the early days of digital audio, the only practical storage device with sufficient bandwidth and storage space was a video recorder and these were adapted to store the digital signal, usually by interfacing said video recorder to a PCM adaptor. Some simple mathematics shows that it is possible to use either 525/60 NTSC or 625/50 PAL video with a sampling rate of 44.1 kHz, a sample rate which persisted with the introduction of CD. A storage device is a device used for storing something. ...
Bandwidth is a measure of frequency range and is typically measured in hertz. ...
Internal Links Related to Video recorder Digital video recorder Video tape recorder Videocassette recorder Video Recordings Act 1984 ...
High-quality PCM audio requires a significantly larger bandwidth than a regular FM audio signal. ...
NTSC is the analog television system in use in Korea, Japan, United States, Canada and certain other places, mostly in the Americas (see map). ...
PAL, short for phase-alternating line, phase alternation by line or phase alternation line, is a colour encoding system used in broadcast television systems in large parts of the world. ...
16 bit digital audio was adopted as the broadcast standard because it offers 96 decibels (dB) of dynamic range, enough to match the quality of broadcast analogue. Modern systems do not suffer as much from the earlier constraints of bandwidth and storage space; 96 kHz and 192 kHz sample rates and 24-bit samples are now common. The sample rate timing can now be quite precise. In computer architecture, 16-bit is an adjective used to describe integers, memory addresses or other data units that are at most 16 bits (2 octets) wide, or to describe CPU and ALU architectures based on registers, address buses, or data buses of that size. ...
The decibel is a dimensionless unit (like percent) that is a measure of ratios on a logarithmic scale. ...
Dynamic range is a term used frequently in numerous fields to describe the ratio between the smallest and largest possible values of a changeable quantity. ...
In computer architecture, 24-bit is an adjective used to describe integers, memory addresses or other data units that are at most 24 bits (3 octets) wide, or to describe CPU and ALU architectures based on registers, address buses, or data buses of that size. ...
Pulse-code Modulation (PCM) is by far the most common way of representing a digital signal. It is simple and is compressed. A PCM representation of an analogue signal is generated by measuring (sampling) the instantaneous amplitude of the analogue signal, and then quantising the result to the nearest bit. However, such rounding contributes to the loss of the original information. Pulse-code modulation (PCM) is a modulation technique. ...
In signal processing, sampling is the reduction of a continuous signal to a discrete signal. ...
Amplitude is a nonnegative scalar measure of a waves magnitude of oscillation, that is, magnitude of the maximum disturbance in the medium during one wave cycle. ...
Quantized signal Digital signal In digital signal processing, quantization is the process of approximating a continuous range of values (or a very large set of possible discrete values) by a relatively-small set of discrete symbols or integer values. ...
Subjective evaluation Whether a sound is "good" or not is subjective -- it can not be easily or objectively measured. It will depend upon the listener's preferences and hearing capabilities, the listener's and the speaker's placement in a given room, and the room's physical properties. The idea is to reproduce the music in such a manner that the sonic and emotional message is faithfully communicated to the listener: for example where replay of a live recording captures the sensation of being at a "live" performance. The arguments are valid for the evaluation of any audio system and not exclusively digital systems. Whilst "controlled" listening tests are difficult, a musician who has played the song, or one who attended several symphony concerts could be a good judge. Of interest are qualities like pitch, echo, spatial origins, tone, timing, phase, excitement/pace, timbre, detail, dynamic range and body.
History of digital audio Commercial digital recording of classical and jazz music began in the early 1970s, pioneered by Japanese companies such as Denon and British record label Decca (who in the mid-70s developed digital audio recorders of their own design for mastering of their albums), although experimental recordings exist from the 1960s. The first 16-bit PCM recording in the United States was made by Thomas Stockham at the Santa Fe Opera in 1976 on a Soundstream recorder. In most cases there was no mixing stage involved; a stereo digital recording was made and used unaltered as the master tape for subsequent commercial release. These unmixed digital recordings are still described as DDD since the technology involved is purely digital. (Unmixed analogue recordings are likewise usually described as ADD to denote a single generation of analogue recording.) Denon is a Japanese electronics company that specializes in high-fidelity audio equipment. ...
It has been suggested that Decca Music Group be merged into this article or section. ...
Thomas Greenway Stockham, (December 22, 1933-January 6, 2004) was an American scientist who developed the first practical digital audio recording system, and pioneered techniques for digital audio recording and processing as well. ...
The Santa Fe Opera is an opera company in Santa Fe, New Mexico. ...
1976 (MCMLXXVI) was a leap year starting on Thursday. ...
Soundstream Inc. ...
SPARS is an acronym for the Society of Professional Audio Recording Studios. ...
SPARS is an acronym for the Society of Professional Audio Recording Studios. ...
The first digitally recorded (DDD) popular music album was Ry Cooder's Bop Till You Drop, recorded in late 1978. It was unmixed, being recorded straight to a two-track 3M digital recorder in the studio. Many other top recording artists were early adherents of digital recording. Stevie Wonder adopted the technology in early 1979 for Journey through the Secret Life of Plants and used it on all later recordings. Others, such as former Beatles producer George Martin, felt that the multitrack digital recording technology of the early 1980s had not reached the sophistication of analogue systems. Martin used digital mixing, however, to eliminate the distortion and noise that an analogue master tape would introduce (thus ADD). An early example of an analogue recording that was digitally mixed is Fleetwood Mac's 1979 release Tusk. Ryland Ry Peter Cooder (born on March 15, 1947) is an American guitarist, singer and composer, known for his slide guitar work, his interest in the American roots music and, more recently, for his collaborations with traditional musicians from many countries. ...
1978 (MCMLXXVIII) was a common year starting on Sunday. ...
3M Company (NYSE: MMM); formerly Minnesota Mining and Manufacturing Company until 2002) is an American corporation with a worldwide presence that produces over 55,000 products, including adhesives, abrasives, laminates, passive fire protection, electronic circuits and displays, and pharmaceuticals. ...
Stevie Wonder (born Stevland Judkins[1] on May 13, 1950), is an American singer, songwriter, record producer, musician, and social activist. ...
This page refers to the year 1979. ...
Journey Through The Secret Life Of Plants was Stevie Wonders most confusing, under-rated and under-appreciated work, out of a nearly 20 year career up to that point. ...
The Beatles appeared on The Ed Sullivan Show in 1964 as part of their first tour of the United States, promoting their first hit single there, I Want To Hold Your Hand. ...
This article or section does not cite its references or sources. ...
This article or section is not written in the formal tone expected of an encyclopedia article. ...
Tusk was a double album released in 1979 (see 1979 in music) by Fleetwood Mac. ...
Digital audio technologies To meet Wikipedias quality standards, this article or section may require cleanup. ...
Digital Audio Broadcasting or DAB is a technology for broadcasting audio programming in digital form that was designed in the late 1980s. ...
To meet Wikipedias quality standards, this article may require cleanup. ...
This article is becoming very long. ...
// Overview The MiniDisc logo A MiniDisc (MD) is a disc-based data storage device intended for storage of digitized audio. ...
Super Audio CD (SACD) is a new audio recording format aimed at providing higher fidelity audio reproduction than the compact disc. ...
A Digital Audio Workstation (DAW) is a system designed to record, edit, and play back digital audio. ...
An embedded hard drive-based player (Apple iPod) A MP3 CD player (Philips Expanium) A flash-based player (iBox Mediaman) A digital audio player (DAP) is a device that stores, organizes and plays digital music files. ...
To meet Wikipedias quality standards, this article or section may require cleanup. ...
Digital audio interfaces Audio signals can also be carried losslessly over general-purpose buses such as USB or FireWire. AC97 (short for Audio Codec 97) is Intel Corporations Audio Codec standard developed by the Intel Architecture Labs in 1997, and used mainly in on-motherboards (a. ...
Integrated circuit showing memory blocks, logic and input/output pads around the periphery A monolithic integrated circuit (also known as IC, microchip, silicon chip, computer chip or chip) is a miniaturized electronic circuit (consisting mainly of semiconductor devices, as well as passive components) which has been manufactured in the surface...
This is an article about the digital recording format. ...
The digital audio standard frequently called AES/EBU, for Audio Engineering Society / European Broadcasting Union, that is officially known as AES3, was first published in 1992 as a standard (and revised in 1995, 1998, and 2003) for carrying digital audio signals between various devices. ...
The XLR connector is a rugged electrical connector design. ...
Background This article describes a standardised method of interconnecting digital audio over a telecommunication standard network. ...
AES/EBU, for Audio Engineering Society / European Broadcasting Union, officially known as AES3, is a 1992 standard (revised in 1995, 1998, and 2003) for carrying digital audio signals between various devices. ...
Asynchronous Transfer Mode (ATM) is a cell relay network protocol which encodes data traffic into small fixed-sized (53 byte; 48 bytes of data and 5 bytes of header information) cells instead of variable sized packets (sometimes known as frames) as in packet-switched networks (such as the Internet Protocol...
I2S, or Inter-IC Sound, is an electrical interface standard used for connecting digital audio devices together. ...
Integrated circuit showing memory blocks, logic and input/output pads around the periphery A monolithic integrated circuit (also known as IC, microchip, silicon chip, computer chip or chip) is a miniaturized electronic circuit (consisting mainly of semiconductor devices, as well as passive components) which has been manufactured in the surface...
Musical Instrument Digital Interface, or MIDI, is a system designed to transmit information between electronic musical instruments. ...
S/PDIF or S/P-DIF stands for Sony/Philips Digital Interface Format, also IEC 958 type II, part of IEC-60958. ...
Radio-grade flexible coaxial cable. ...
TOSLINK connector (JIS F05) TOSLINK⢠is a standardized optical fibre connection system. ...
The USB (Type A and B) Connectors A USB Series âAâ plug USB Logo High Speed USB Logo USB OTG Logo Wireless USB Logo Universal Serial Bus (USB) is a serial bus standard to interface devices. ...
FireWire (also known as i. ...
References - Borwick, John, ed., 1994: Sound Recording Practice (Oxford: Oxford University Press)
- Ifeachor, Emmanuel C., and Jervis, Barrie W., 2002: Digital Signal Processing: A Practical Approach (Harlow, England: Pearson Education Limited)
- Rabiner, Lawrence R., and Gold, Bernard, 1975: Theory and Application of Digital Signal Processing (Englewood Cliffs, New Jersey: Prentice-Hall, Inc.)
- Watkinson, John, 1994: The Art of Digital Audio (Oxford: Focal Press)
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