Digital audio is a representation of sound recorded in, or converted into, digital form . In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio , samples are taken 44,100 times per second , each with 16-bit resolution . Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced analog audio technology in many areas of audio engineering , record production and telecommunications in the 1990s and 2000s.
47-504: The ADAT Lightpipe , officially the ADAT Optical Interface , is a standard for the transfer of digital audio between equipment. It was originally developed by Alesis but has since become widely accepted, with many third party hardware manufacturers including Lightpipe interfaces on their equipment. The protocol has become so popular that the term ADAT is now often used to refer to the transfer standard rather than to
94-422: A data compression algorithm. Adaptive DPCM (ADPCM) was introduced by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973. Perceptual coding was first used for speech coding compression, with linear predictive coding (LPC). Initial concepts for LPC date back to the work of Fumitada Itakura ( Nagoya University ) and Shuzo Saito ( Nippon Telegraph and Telephone ) in 1966. During
141-462: A digital system do not result in error unless they are so large as to result in a symbol being misinterpreted as another symbol or disturbing the sequence of symbols. It is, therefore, generally possible to have an entirely error-free digital audio system in which no noise or distortion is introduced between conversion to digital format and conversion back to analog. A digital audio signal may be encoded for correction of any errors that might occur in
188-486: A digital-to-analog converter (DAC) performs the reverse process, converting a digital signal back into an analog signal, which is then sent through an audio power amplifier and ultimately to a loudspeaker . Digital audio systems may include compression , storage , processing , and transmission components. Conversion to a digital format allows convenient manipulation, storage, transmission, and retrieval of an audio signal. Unlike analog audio, in which making copies of
235-438: A 160 kbit/s raw data rate operates at 120 kBd.) Codes with many symbols, and thus a bit rate higher than the symbol rate, are most useful on channels such as telephone lines with a limited bandwidth but a high signal-to-noise ratio within that bandwidth. In other applications, the bit rate is less than the symbol rate. Eight-to-fourteen modulation as used on audio CDs has bit rate 8 / 17 of
282-489: A DAC. According to the Nyquist–Shannon sampling theorem , with some practical and theoretical restrictions, a band-limited version of the original analog signal can be accurately reconstructed from the digital signal. During conversion, audio data can be embedded with a digital watermark to prevent piracy and unauthorized use. Watermarking is done using a direct-sequence spread-spectrum (DSSS) method. The audio information
329-537: A different sampling rate to a common sampling rate prior to processing. Audio data compression techniques, such as MP3 , Advanced Audio Coding (AAC), Opus , Ogg Vorbis , or FLAC , are commonly employed to reduce the file size. Digital audio can be carried over digital audio interfaces such as AES3 or MADI . Digital audio can be carried over a network using audio over Ethernet , audio over IP or other streaming media standards and systems. For playback, digital audio must be converted back to an analog signal with
376-581: A measure of the gross bit rate R as Here, the ⌈ x ⌉ {\displaystyle \left\lceil x\right\rceil } denotes the ceiling function of x {\displaystyle x} , where x {\displaystyle x} is taken to be any real number greater than zero, then the ceiling function rounds up to the nearest natural number (e.g. ⌈ 2.11 ⌉ = 3 {\displaystyle \left\lceil 2.11\right\rceil =3} ). In that case, M = 2 different symbols are used. In
423-449: A modem, these may be time-limited sinewave tones with unique combinations of amplitude, phase and/or frequency. For example, in a 64QAM modem, M = 64 , and so the bit rate is N = log 2 (64) = 6 times the baud rate. In a line code, these may be M different voltage levels. The ratio is not necessarily an integer; in 4B3T coding, the bit rate is 4 / 3 of the baud rate. (A typical basic rate interface with
470-494: A network solution, offer additional flexibility compared to point-to-point technologies such as Lightpipe. Digital audio In a digital audio system, an analog electrical signal representing the sound is converted with an analog-to-digital converter (ADC) into a digital signal, typically using pulse-code modulation (PCM). This digital signal can then be recorded, edited, modified, and copied using computers , audio playback machines, and other digital tools. For playback,
517-522: A range of digital transmission applications such as the integrated services digital network (ISDN), cordless telephones and cell phones . Digital audio is used in broadcasting of audio. Standard technologies include Digital audio broadcasting (DAB), Digital Radio Mondiale (DRM), HD Radio and In-band on-channel (IBOC). Digital audio in recording applications is stored on audio-specific technologies including CD, DAT, Digital Compact Cassette (DCC) and MiniDisc . Digital audio may be stored in
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#1732787158515564-580: A recording results in generation loss and degradation of signal quality, digital audio allows an infinite number of copies to be made without any degradation of signal quality. Digital audio technologies are used in the recording, manipulation, mass-production, and distribution of sound, including recordings of songs , instrumental pieces, podcasts , sound effects, and other sounds. Modern online music distribution depends on digital recording and data compression . The availability of music as data files, rather than as physical objects, has significantly reduced
611-521: A similar function with Hi8 tapes. Formats like ProDigi and DASH were referred to as SDAT (stationary-head digital audio tape) formats, as opposed to formats like the PCM adaptor-based systems and Digital Audio Tape (DAT), which were referred to as RDAT (rotating-head digital audio tape) formats, due to their helical-scan process of recording. Like the DAT cassette, ProDigi and DASH machines also accommodated
658-442: A specified sampling rate and converts at a known bit resolution. CD audio , for example, has a sampling rate of 44.1 kHz (44,100 samples per second), and has 16-bit resolution for each stereo channel. Analog signals that have not already been bandlimited must be passed through an anti-aliasing filter before conversion, to prevent the aliasing distortion that is caused by audio signals with frequencies higher than
705-405: A standard audio file formats and stored on a Hard disk recorder , Blu-ray or DVD-Audio . Files may be played back on smartphones, computers or MP3 player . Digital audio resolution is measured in audio bit depth . Most digital audio formats use either 16-bit, 24-bit, and 32-bit resolution. Baud rate In telecommunications and electronics , baud ( / b ɔː d / ; symbol: Bd )
752-456: A symbol may have more than two states, so it may represent more than one bit . A bit (binary digit) always represents one of two states. If N bits are conveyed per symbol, and the gross bit rate is R , inclusive of channel coding overhead, the symbol rate f s can be calculated as By taking information per pulse N in bit/pulse to be the base-2- logarithm of the number of distinct messages M that could be sent, Hartley constructed
799-725: Is AES3 , developed by the Audio Engineering Society and the European Broadcasting Union , which transmits two channels of digital audio up to 24-bits 192 kHz over a balanced XLR cable. S/PDIF (Sony/Philips Digital Interface) is the consumer version of this protocol, which uses either RCA leads or optical cables identical to lightpipe cables. MADI can carry 64 channels of audio at 48 kHz, 32 channels at 96 kHz or 16 channels at 192 kHz. Audio over Ethernet and audio over IP use standard network technologies and equipment and, as
846-427: Is a common unit of measurement of symbol rate , which is one of the components that determine the speed of communication over a data channel . It is the unit for symbol rate or modulation rate in symbols per second or pulses per second . It is the number of distinct symbol changes (signalling events) made to the transmission medium per second in a digitally modulated signal or a bd rate line code . Baud
893-453: Is related to gross bit rate , which can be expressed in bits per second (bit/s). If there are precisely two symbols in the system (typically 0 and 1), then baud and bits per second are equivalent. The baud unit is named after Émile Baudot , the inventor of the Baudot code for telegraphy , and is represented according to the rules for SI units . That is, the first letter of its symbol
940-409: Is related to gross bit rate expressed in bit/s. The term baud has sometimes incorrectly been used to mean bit rate , since these rates are the same in old modems as well as in the simplest digital communication links using only one bit per symbol, such that binary digit "0" is represented by one symbol, and binary digit "1" by another symbol. In more advanced modems and data transmission techniques,
987-439: Is reversed for reproduction: the electrical audio signal is amplified and then converted back into physical waveforms via a loudspeaker . Analog audio retains its fundamental wave-like characteristics throughout its storage, transformation, duplication, and amplification. Analog audio signals are susceptible to noise and distortion, due to the innate characteristics of electronic circuits and associated devices. Disturbances in
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#17327871585151034-545: Is sent at the desired sample rate, for a bit rate of 256×48 kHz = 12.288 Mbit/s. This is twice the baud rate used by S/PDIF (3.072 Mbit/s, doubled by biphase coding to 6.144 MBd), but still within the specified 15 Mbaud capacity of the popular TOTX147 [3] /TORX147 [4] TOSLINK transceivers. User data bit allocations: The transmission speed of the user bits is equal to the sampling rate (e.g. 48,000 bits per second) There are numerous digital audio transfer protocols. The most commonly used professional interface
1081-469: Is then modulated by a pseudo-noise (PN) sequence, then shaped within the frequency domain and put back in the original signal. The strength of the embedding determines the strength of the watermark on the audio data. Pulse-code modulation (PCM) was invented by British scientist Alec Reeves in 1937. In 1950, C. Chapin Cutler of Bell Labs filed the patent on differential pulse-code modulation (DPCM),
1128-462: Is turned into an electronic data stream going to an IC chip commonly referred to at Alesis as "the 1-K chip". From there the audio data frame is routed to processing ICs. With an ADAT Lightpipe and an ADAT controller linking up to four ADATs using CAT5 cables with RJ-connectors and SMPTE Time Code, you could synchronize four 8-track ADATs together for a total of 32 simultaneous synchronized channel tracks of 16 or 20 bit audio data. 24 bit came later with
1175-571: Is uppercase (Bd), but when the unit is spelled out, it should be written in lowercase (baud) except when it begins a sentence or is capitalized for another reason, such as in title case. It was defined by the CCITT (now the ITU ) in November 1926. The earlier standard had been the number of words per minute, which was a less robust measure since word length can vary. The symbol duration time, also known as
1222-833: The Alesis Digital Audio Tape itself. Lightpipe uses the same connection hardware as S/PDIF : fiber optic cables (hence its name) to carry data, with Toslink connectors and optical transceivers at either end. However, the data streams of the two protocols are incompatible. S/PDIF is mostly used for transferring stereo or multi-channel surround sound audio, whereas the ADAT optical interface supports up to 8 audio channels at 48 kHz, 24 bit. Lightpipe devices have been successfully interfaced via FireWire . Lightpipe can carry eight channels of uncompressed digital audio at 24 bit resolution at 48,000 samples or four channels at 96,000 samples per second. Initially used for
1269-552: The Nyquist frequency (half the sampling rate). A digital audio signal may be stored or transmitted. Digital audio can be stored on a CD, a digital audio player , a hard drive , a USB flash drive , or any other digital data storage device . The digital signal may be altered through digital signal processing , where it may be filtered or have effects applied. Sample-rate conversion including upsampling and downsampling may be used to change signals that have been encoded with
1316-720: The United States was made by Thomas Stockham at the Santa Fe Opera in 1976, on a Soundstream recorder. An improved version of the Soundstream system was used to produce several classical recordings by Telarc in 1978. The 3M digital multitrack recorder in development at the time was based on BBC technology. The first all-digital album recorded on this machine was Ry Cooder 's Bop till You Drop in 1979. British record label Decca began development of its own 2-track digital audio recorders in 1978 and released
1363-399: The unit interval , can be directly measured as the time between transitions by looking at an eye diagram of the signal on an oscilloscope . The symbol duration time T s can be calculated as: where f s is the symbol rate. There is also a chance of miscommunication which leads to ambiguity. The baud is scaled using standard metric prefixes , so that for example The symbol rate
1410-434: The 1970s, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs developed a form of LPC called adaptive predictive coding (APC), a perceptual coding algorithm that exploited the masking properties of the human ear, followed in the early 1980s with the code-excited linear prediction (CELP) algorithm. Discrete cosine transform (DCT) coding, a lossy compression method first proposed by Nasir Ahmed in 1972, provided
1457-548: The HD24 hard disk recorder in early 2001, which also made use of Lightpipe capabilities. The lightpipe is "hot-pluggable", which means devices do not need to be turned off for plugging in or unplugging (although it is advisable to mute the receiving equipment, since there will be a large signal spike when the connection is made). The optical connect avoids ground-loops, which can be troublesome in larger installations, and will not transfer any harmful electrical spikes from one device to
ADAT Lightpipe - Misplaced Pages Continue
1504-458: The Lightpipe format was modified using bit-splitting techniques by the company Sonorus. Known as S/MUX (short for 'sample multiplexing'), this connection allows 4 channels at up to 96 kHz, or two channels at up to 192 kHz, on one optical cable. Most manufacturers implementing ADAT Lightpipe now support this S/MUX interface extension. Light carrying the data signal through the Lightpipe
1551-490: The audio data being recorded to the tape using a multi-track stationary tape head. PCM adaptors allowed for stereo digital audio recording on a conventional NTSC or PAL video tape recorder . The 1982 introduction of the CD by Philips and Sony popularized digital audio with consumers. ADAT became available in the early 1990s, which allowed eight-track 44.1 or 48 kHz recording on S-VHS cassettes, and DTRS performed
1598-419: The audio information while the other eight are simply occupied by zeros. The receiving device ignores information it cannot process. For example, a 20 bit signal going from a Type II ADAT to a Type I (which only operates at 16 bits) will simply ignore the bits below the sixteen MSBs. Higher sample rates can be accommodated with a reduced number of channels. While the original ADAT machines did not support this,
1645-466: The basis for the modified discrete cosine transform (MDCT), which was developed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987. The MDCT is the basis for most audio coding standards , such as Dolby Digital (AC-3), MP3 ( MPEG Layer III), AAC, Windows Media Audio (WMA), Opus and Vorbis ( Ogg ). PCM was used in telecommunications applications long before its first use in commercial broadcast and recording. Commercial digital recording
1692-419: The bitstream is not biphase mark coded like S/PDIF. Instead, NRZI coding is used, where a 0 bit indicates no transition and a 1 bit is a transition. 8 audio samples at 24 bits per sample plus 4 user bits (196 bits total) are sent in groups of 4 data bits followed by a 1 bit to force a transition. This totals 196×5/4 = 245 bits. 10 consecutive 0 bits followed by a 1 bit provide frame synchronization. One frame
1739-482: The computer can effectively run at a single time. Avid Audio and Steinberg released the first digital audio workstation software programs in 1989. Digital audio workstations make multitrack recording and mixing much easier for large projects which would otherwise be difficult with analog equipment. The rapid development and wide adoption of PCM digital telephony was enabled by metal–oxide–semiconductor (MOS) switched capacitor (SC) circuit technology, developed in
1786-535: The costs of distribution as well as making it easier to share copies. Before digital audio, the music industry distributed and sold music by selling physical copies in the form of records and cassette tapes . With digital audio and online distribution systems such as iTunes , companies sell digital sound files to consumers, which the consumer receives over the Internet. Popular streaming services such as Apple Music , Spotify , or YouTube , offer temporary access to
1833-399: The digital file, and are now the most common form of music consumption. An analog audio system converts physical waveforms of sound into electrical representations of those waveforms by use of a transducer , such as a microphone . The sounds are then stored on an analog medium such as magnetic tape , or transmitted through an analog medium such as a telephone line or radio . The process
1880-570: The early 1970s. This led to the development of PCM codec-filter chips in the late 1970s. The silicon-gate CMOS (complementary MOS) PCM codec-filter chip, developed by David A. Hodges and W.C. Black in 1980, has since been the industry standard for digital telephony. By the 1990s, telecommunication networks such as the public switched telephone network (PSTN) had been largely digitized with VLSI (very large-scale integration ) CMOS PCM codec-filters, widely used in electronic switching systems for telephone exchanges , user-end modems and
1927-399: The first European digital recording in 1979. Popular professional digital multitrack recorders produced by Sony/Studer ( DASH ) and Mitsubishi ( ProDigi ) in the early 1980s helped to bring about digital recording's acceptance by the major record companies. Machines for these formats had their own transports built-in as well, using reel-to-reel tape in either 1/4", 1/2", or 1" widths, with
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1974-460: The next. Lightpipe was designed for use with the Alesis ADATs, and although extremely versatile, there are a few limitations. For straightforward digital audio transfer, the receiving device can synchronize to the lightpipe's embedded clock signal, achieving a 1:1 digital copy. For transport control, additional synchronization is needed between devices. (For example, using two ADAT machines at
2021-578: The obligatory 44.1 kHz sampling rate, but also 48 kHz on all machines, and eventually a 96 kHz sampling rate. They overcame the problems that made typical analog recorders unable to meet the bandwidth (frequency range) demands of digital recording by a combination of higher tape speeds, narrower head gaps used in combination with metal-formulation tapes, and the spreading of data across multiple parallel tracks. Unlike analog systems, modern digital audio workstations and audio interfaces allow as many channels in as many different sampling rates as
2068-429: The same time to achieve 16-channel throughput would require better transport control; otherwise, the two ADAT machines would be very unlikely to play in sync.) Nine pin D connectors are used to transfer transport information. The Alesis ADAT HD24 also offers MIDI Time Code for synchronization with MIDI-enabled devices. In order to fit 8 channels within the bandwidth limits of the standard TOSLINK transceiver modules,
2115-400: The storage or transmission of the signal. This technique, known as channel coding , is essential for broadcast or recorded digital systems to maintain bit accuracy. Eight-to-fourteen modulation is the channel code used for the audio compact disc (CD). If an audio signal is analog, a digital audio system starts with an ADC that converts an analog signal to a digital signal. The ADC runs at
2162-472: The transfer of digital audio between ADATs, the protocol was designed with future improvements in mind. All Lightpipe signals are transmitted at 24 bit resolution, no matter what the depth of the audio; information is contained within the Most Significant Bits and the rest of the bits remain a string of zeros. For example, if a 16 bit signal is sent via Lightpipe, the first sixteen bits contain
2209-499: Was pioneered in Japan by NHK and Nippon Columbia and their Denon brand, in the 1960s. The first commercial digital recordings were released in 1971. The BBC also began to experiment with digital audio in the 1960s. By the early 1970s, it had developed a 2-channel recorder, and in 1972 it deployed a digital audio transmission system that linked their broadcast center to their remote transmitters. The first 16-bit PCM recording in
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