Digital audio

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Audio levels display on a digital audio recorder (Zoom H4n) Zoom H4n audio recording levels.jpg
Audio levels display on a digital audio recorder (Zoom H4n)

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.

Contents

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, 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 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.

Overview

A sound wave, in red, represented digitally, in blue (after sampling and 4-bit quantization). 4-bit-linear-PCM.svg
A sound wave, in red, represented digitally, in blue (after sampling and 4-bit quantization).

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 the costs of distribution as well as making it easier to share copies. [1] 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 the digital file, and are now the most common form of music consumption. [2]

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 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 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]

A digital audio signal may be encoded for correction of any errors that might occur in 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).

Conversion process

The lifecycle of sound from its source, through an ADC, digital processing, a DAC, and finally as sound again. A-D-A Flow.svg
The lifecycle of sound from its source, through an ADC, digital processing, a DAC, and finally as sound again.

If an audio signal is analog, a digital audio system starts with an ADC that converts an analog signal to a digital signal. [b] The ADC runs at 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 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 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 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 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. [4]

History

Coding

Pulse-code modulation (PCM) was invented by British scientist Alec Reeves in 1937. [5] In 1950, C. Chapin Cutler of Bell Labs filed the patent on differential pulse-code modulation (DPCM), [6] a data compression algorithm. Adaptive DPCM (ADPCM) was introduced by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973. [7] [8]

Perceptual coding was first used for speech coding compression, with linear predictive coding (LPC). [9] Initial concepts for LPC date back to the work of Fumitada Itakura (Nagoya University) and Shuzo Saito (Nippon Telegraph and Telephone) in 1966. [10] During 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. [9]

Discrete cosine transform (DCT) coding, a lossy compression method first proposed by Nasir Ahmed in 1972, [11] [12] provided 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. [13] The MDCT is the basis for most audio coding standards, such as Dolby Digital (AC-3), [14] MP3 (MPEG Layer III), [15] [9] AAC, Windows Media Audio (WMA), Opus and Vorbis (Ogg). [14]

Recording

Analog reel-to-reel tape recorder Reel to reel Hitachi I (1972).JPG
Analog reel-to-reel tape recorder
Sony professional digital audio tape (DAT) recorder PCM-7030 Sony PCM-7030 of DR 20111102a-crop.jpg
Sony professional digital audio tape (DAT) recorder PCM-7030
Digital audio workstation REAPER Digital Audio Workstation.jpg
Digital audio workstation

PCM was used in telecommunications applications long before its first use in commercial broadcast and recording. Commercial digital recording 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. [16]

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. [16]

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. 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 the first European digital recording in 1979. [16]

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 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. [16]

ADAT became available in the early 1990s, which allowed eight-track 44.1 or 48 kHz recording on S-VHS cassettes, and DTRS performed 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 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 the computer can effectively run at a single time. Avid Audio and Steinberg released the first digital audio workstation software programs in 1989. [17] Digital audio workstations make multitrack recording and mixing much easier for large projects which would otherwise be difficult with analog equipment.

Telephony

The rapid development and wide adoption of PCM digital telephony was enabled by metal–oxide–semiconductor (MOS) switched capacitor (SC) circuit technology, developed in the early 1970s. [18] This led to the development of PCM codec-filter chips in the late 1970s. [18] [19] The silicon-gate CMOS (complementary MOS) PCM codec-filter chip, developed by David A. Hodges and W.C. Black in 1980, [18] has since been the industry standard for digital telephony. [18] [19] 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 a range of digital transmission applications such as the integrated services digital network (ISDN), cordless telephones and cell phones. [19]

Technologies

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 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.

See also

Notes

  1. ↑ Anti-alias filtering and optional digital signal processing may degrade the audio signal via passband ripple, non-linear phase shift, numeric precision quantization noise or time distortion of transients. However, these potential degradations can be limited by careful digital design. [3]
  2. ↑ Some audio signals such as those created by digital synthesis originate entirely in the digital domain, in which case analog to digital conversion does not take place.

Related Research Articles

Audio signal processing is a subfield of signal processing that is concerned with the electronic manipulation of audio signals. Audio signals are electronic representations of sound waves—longitudinal waves which travel through air, consisting of compressions and rarefactions. The energy contained in audio signals or sound power level is typically measured in decibels. As audio signals may be represented in either digital or analog format, processing may occur in either domain. Analog processors operate directly on the electrical signal, while digital processors operate mathematically on its digital representation.

In electronics and telecommunications, modulation is the process of varying one or more properties of a periodic waveform, called the carrier signal, with a separate signal called the modulation signal that typically contains information to be transmitted. For example, the modulation signal might be an audio signal representing sound from a microphone, a video signal representing moving images from a video camera, or a digital signal representing a sequence of binary digits, a bitstream from a computer.

<span class="mw-page-title-main">Analog-to-digital converter</span> System that converts an analog signal into a digital signal

In electronics, an analog-to-digital converter is a system that converts an analog signal, such as a sound picked up by a microphone or light entering a digital camera, into a digital signal. An ADC may also provide an isolated measurement such as an electronic device that converts an analog input voltage or current to a digital number representing the magnitude of the voltage or current. Typically the digital output is a two's complement binary number that is proportional to the input, but there are other possibilities.

<span class="mw-page-title-main">Delta modulation</span> Signal conversion technique

Delta modulation is an analog-to-digital and digital-to-analog signal conversion technique used for transmission of voice information where quality is not of primary importance. DM is the simplest form of differential pulse-code modulation (DPCM) where the difference between successive samples is encoded into n-bit data streams. In delta modulation, the transmitted data are reduced to a 1-bit data stream representing either up (↗) or down (↘). Its main features are:

<span class="mw-page-title-main">Digital Audio Tape</span> Digital audio cassette format developed by Sony

Digital Audio Tape is a signal recording and playback medium developed by Sony and introduced in 1987. In appearance it is similar to a Compact Cassette, using 3.81 mm / 0.15" magnetic tape enclosed in a protective shell, but is roughly half the size at 73 mm × 54 mm × 10.5 mm. The recording is digital rather than analog. DAT can record at sampling rates equal to, as well as higher and lower than a CD at 16 bits quantization. If a comparable digital source is copied without returning to the analogue domain, then the DAT will produce an exact clone, unlike other digital media such as Digital Compact Cassette or non-Hi-MD MiniDisc, both of which use a lossy data-reduction system.

Sound can be recorded and stored and played using either digital or analog techniques. Both techniques introduce errors and distortions in the sound, and these methods can be systematically compared. Musicians and listeners have argued over the superiority of digital versus analog sound recordings. Arguments for analog systems include the absence of fundamental error mechanisms which are present in digital audio systems, including aliasing and associated anti-aliasing filter implementation, jitter and quantization noise. Advocates of digital point to the high levels of performance possible with digital audio, including excellent linearity in the audible band and low levels of noise and distortion.

<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).

<span class="mw-page-title-main">Digital recording</span> Audio or video represented as a stream of discrete numbers

In digital recording, an audio or video signal is converted into a stream of discrete numbers representing the changes over time in air pressure for audio, or chroma and luminance values for video. This number stream is saved to a storage device. To play back a digital recording, the numbers are retrieved and converted back into their original analog audio or video forms so that they can be heard or seen.

The digital sound revolution refers to the widespread adoption of digital audio technology in the computer industry beginning in the 1980s.

<span class="mw-page-title-main">Delta-sigma modulation</span> Method for converting signals between digital and analog

Delta-sigma modulation is an oversampling method for encoding signals into low bit depth digital signals at a very high sample-frequency as part of the process of delta-sigma analog-to-digital converters (ADCs) and digital-to-analog converters (DACs). Delta-sigma modulation achieves high quality by utilizing a negative feedback loop during quantization to the lower bit depth that continuously corrects quantization errors and moves quantization noise to higher frequencies well above the original signal's bandwidth. Subsequent low-pass filtering for demodulation easily removes this high frequency noise and time averages to achieve high accuracy in amplitude which can be ultimately encoded as pulse-code modulation (PCM).

<span class="mw-page-title-main">PCM adaptor</span> Encodes digital audio as video

A PCM adaptor is a device that encodes digital audio as video for recording on a videocassette recorder. The adapter also has the ability to decode a video signal back to digital audio for playback. This digital audio system was used for mastering early compact discs.

Soundstream Inc. was the first United States audiophile digital audio recording company, providing commercial services for recording and computer-based editing.

The dbx Model 700 Digital Audio Processor was a professional audio ADC/DAC combination unit, which digitized a stereo analog audio input into a bitstream, which was then encoded and encapsulated in an analog composite video signal, for recording to tape using a VCR as a transport. Unlike other similar pieces of equipment like the Sony PCM-F1, the Model 700 used a technique called Companded Predictive Delta Modulation, rather than the now-common pulse-code modulation. At the time of its introduction in the mid-1980s the device was the first commercial product to use this method, although it had been proposed in the 1960s and prototyped in the late '70s.

<span class="mw-page-title-main">Audio bit depth</span> Number of bits of information recorded for each digital audio sample

In digital audio using pulse-code modulation (PCM), bit depth is the number of bits of information in each sample, and it directly corresponds to the resolution of each sample. Examples of bit depth include Compact Disc Digital Audio, which uses 16 bits per sample, and DVD-Audio and Blu-ray Disc, which can support up to 24 bits per sample.

A Bitcrusher is an audio effect that produces distortion by reducing the resolution or bandwidth of digital audio data. The resulting quantization noise may produce a "warmer" sound impression, or a harsh one, depending on the amount of reduction.

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 the required data bandwidth for a given signal-to-noise ratio.

<span class="mw-page-title-main">Sub-band coding</span>

In signal processing, sub-band coding (SBC) is any form of transform coding that breaks a signal into a number of different frequency bands, typically by using a fast Fourier transform, and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.

Pulse-code modulation (PCM) is a method used to digitally represent 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 at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Alec Reeves, Claude Shannon, Barney Oliver and John R. Pierce are credited with its invention.

<span class="mw-page-title-main">Audio coding format</span> Digitally coded format for audio signals

An audio coding format is a content representation format for storage or transmission of digital audio. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.

References

  1. ↑ Janssens, Jelle; Stijn Vandaele; Tom Vander Beken (2009). "The Music Industry on (the) Line? Surviving Music Piracy in a Digital Era". European Journal of Crime, Criminal Law and Criminal Justice. 77 (96): 77–96. doi:10.1163/157181709X429105. hdl: 1854/LU-608677 .
  2. ↑ Liikkanen, Lassi A.; Åman, Pirkka (May 2016). "Shuffling Services: Current Trends in Interacting with Digital Music". Interacting with Computers. 28 (3): 352–371. doi:10.1093/iwc/iwv004. ISSN   0953-5438.
  3. ↑ Story, Mike (September 1997). "A Suggested Explanation For (Some Of) The Audible Differences Between High Sample Rate And Conventional Sample Rate Audio Material" (PDF). dCS Ltd. Archived (PDF) from the original on 28 November 2009.
  4. ↑ Seok, Jongwon; Hong, Jinwoo; Kim, Jinwoong (2002-06-01). "A Novel Audio Watermarking Algorithm for Copyright Protection of Digital Audio". ETRI Journal. 24 (3): 181–189. doi: 10.4218/etrij.02.0102.0301 . ISSN   1225-6463. S2CID   3008374.
  5. ↑ Genius Unrecognised, BBC, 2011-03-27, retrieved 2011-03-30
  6. ↑ USpatent 2605361,C. Chapin Cutler,"Differential Quantization of Communication Signals",issued 1952-07-29
  7. ↑ P. Cummiskey, Nikil S. Jayant, and J. L. Flanagan, "Adaptive quantization in differential PCM coding of speech", Bell Syst. Tech. J., vol. 52, pp. 1105—1118, Sept. 1973
  8. ↑ Cummiskey, P.; Jayant, Nikil S.; Flanagan, J. L. (1973). "Adaptive quantization in differential PCM coding of speech". The Bell System Technical Journal. 52 (7): 1105–1118. doi:10.1002/j.1538-7305.1973.tb02007.x. ISSN   0005-8580.
  9. 1 2 3 Schroeder, Manfred R. (2014). "Bell Laboratories". Acoustics, Information, and Communication: Memorial Volume in Honor of Manfred R. Schroeder. Springer. p. 388. ISBN   9783319056609.
  10. ↑ Gray, Robert M. (2010). "A History of Realtime Digital Speech on Packet Networks: Part II of Linear Predictive Coding and the Internet Protocol" (PDF). Found. Trends Signal Process. 3 (4): 203–303. doi: 10.1561/2000000036 . ISSN   1932-8346.
  11. ↑ Ahmed, Nasir (January 1991). "How I Came Up With the Discrete Cosine Transform". Digital Signal Processing . 1 (1): 4–5. Bibcode:1991DSP.....1....4A. doi:10.1016/1051-2004(91)90086-Z.
  12. ↑ Nasir Ahmed; T. Natarajan; Kamisetty Ramamohan Rao (January 1974). "Discrete Cosine Transform" (PDF). IEEE Transactions on Computers. C-23 (1): 90–93. doi:10.1109/T-C.1974.223784. S2CID   149806273.
  13. ↑ J. P. Princen, A. W. Johnson und A. B. Bradley: Subband/transform coding using filter bank designs based on time domain aliasing cancellation, IEEE Proc. Intl. Conference on Acoustics, Speech, and Signal Processing (ICASSP), 2161–2164, 1987.
  14. 1 2 Luo, Fa-Long (2008). Mobile Multimedia Broadcasting Standards: Technology and Practice. Springer Science & Business Media. p. 590. ISBN   9780387782638.
  15. ↑ Guckert, John (Spring 2012). "The Use of FFT and MDCT in MP3 Audio Compression" (PDF). University of Utah . Retrieved 14 July 2019.
  16. 1 2 3 4 Fine, Thomas (2008). Barry R. Ashpole (ed.). "The Dawn of Commercial Digital Recording" (PDF). ARSC Journal. Retrieved 2010-05-02.
  17. ↑ Reuter, Anders (2022-03-15). "Who let the DAWs Out? The Digital in a New Generation of the Digital Audio Workstation". Popular Music and Society. 45 (2): 113–128. doi:10.1080/03007766.2021.1972701. ISSN   0300-7766. S2CID   242779244.
  18. 1 2 3 4 Allstot, David J. (2016). "Switched Capacitor Filters" (PDF). In Maloberti, Franco; Davies, Anthony C. (eds.). A Short History of Circuits and Systems: From Green, Mobile, Pervasive Networking to Big Data Computing. IEEE Circuits and Systems Society. pp. 105–110. ISBN   9788793609860. Archived from the original (PDF) on 2021-09-30. Retrieved 2019-11-29.
  19. 1 2 3 Floyd, Michael D.; Hillman, Garth D. (8 October 2018) [1st pub. 2000]. "Pulse-Code Modulation Codec-Filters". The Communications Handbook (2nd ed.). CRC Press. pp. 26–1, 26–2, 26–3. ISBN   9781420041163.

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