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96-454: The audio data in most AIFF files is uncompressed pulse-code modulation (PCM). This type of AIFF file uses much more disk space than lossy formats like MP3 —about 10 MB for one minute of stereo audio at a sample rate of 44.1 kHz and a bit depth of 16 bits. There is also a compressed variant of AIFF known as AIFF-C or AIFC , with various defined compression codecs. In addition to audio data, AIFF can include loop point data and

192-422: 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

288-462: 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

384-526: A finite impulse response filter but has no practical implementation using analog components. A practical advantage of digital processing is the more convenient recall of settings. Plug-in parameters can be stored on the computer, whereas parameter details on an analog unit must be written down or otherwise recorded if the unit needs to be reused. This can be cumbersome when entire mixes must be recalled manually using an analog console and outboard gear. When working digitally, all parameters can simply be stored in

480-454: A 16-bit analog-to-digital converter may have a dynamic range of between 90 and 95 dB, whereas the signal-to-noise ratio (roughly the equivalent of dynamic range, noting the absence of quantization noise but presence of tape hiss) of a professional reel-to-reel ¼-inch tape recorder would be between 60 and 70 dB at the recorder's rated output. The benefits of using digital recorders with greater than 16-bit accuracy can be applied to

576-498: A 16-bit system would be sufficient, but noted the small reserve the system provided in ordinary operating conditions. For this reason, it was suggested that a fast-acting signal limiter or ' soft clipper ' be used to prevent the system from becoming overloaded. With many recordings, high level distortions at signal peaks may be audibly masked by the original signal, thus large amounts of distortion may be acceptable at peak signal levels. The difference between analog and digital systems

672-489: 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

768-460: A DAW project file and recalled instantly. Most modern professional DAWs also process plug-ins in real time, which means that processing can be largely non-destructive until final mix-down. Many plug-ins exist now that incorporate analog modeling. There are audio engineers that endorse them and feel that they compare equally in sound to the analog processes that they imitate. Analog modeling carries some benefits over their analog counterparts, such as

864-488: A certain amount of time for listener training. Early digital audio machines had disappointing results, with digital converters introducing errors that the ear could detect. Record companies released their first LPs based on digital audio masters in the late 1970s. CDs became available in the early 1980s. At this time analog sound reproduction was a mature technology . There was a mixed critical response to early digital recordings released on CD. Compared to vinyl record, it

960-549: 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

1056-537: 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

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1152-462: A digital filter. This approach has several advantages since the digital filter can be made to have a near-ideal frequency domain transfer function, with low in-band ripple, and no aging or thermal drift. However, the digital anti-aliasing filter may introduce degradations due to time domain response particularly at lower sample rates. Analog systems are not subject to a Nyquist limit or aliasing and thus do not require anti-aliasing filters or any of

1248-400: A dynamic range of up to 77 dB. An LP made out of perfect diamond has an atomic feature size of about 0.5 nanometer , which, with a groove size of 8 micron , yields a theoretical dynamic range of 110 dB. An LP made out of perfect vinyl LP would have a theoretical dynamic range of 70 dB. Measurements indicate maximum actual performance in the 60 to 70 dB range. Typically,

1344-542: A few of the 176 test subjects. A perceptual study by Nishiguchi et al. (2004) concluded that "no significant difference was found between sounds with and without very high frequency components among the sound stimuli and the subjects... however, [Nishiguchi et al] can still neither confirm nor deny the possibility that some subjects could discriminate between musical sounds with and without very high frequency components." In blind listening tests conducted by Bob Katz in 1996, recounted in his book Mastering Audio: The Art and

1440-424: A groove size of 8 micron and a feature size of 0.5 nanometer, has a quantization that is similar to a 16-bit digital sample. It is possible to make quantization noise audibly benign by applying dither . To do this, noise is added to the original signal before quantization. Optimal use of dither has the effect of making quantization error independent of the signal, and allows signal information to be retained below

1536-584: A laser beam. Therefore, no such media deterioration takes place, and the CD will, with proper care, sound exactly the same every time it is played (discounting aging of the player and CD itself); however, this is a benefit of the optical system, not of digital recording, and the Laserdisc format enjoys the same non-contact benefit with analog optical signals. CDs suffer from disc rot and slowly degrade with time, even if they are stored properly and not played. M-DISC ,

1632-422: A less aggressive and lower-cost anti-aliasing filter to be used. Early digital systems may have suffered from a number of signal degradations related to the use of analog anti-aliasing filters, e.g., time dispersion, nonlinear distortion , ripple , temperature dependence of filters etc. Using an oversampling design and delta-sigma modulation , a less aggressive analog anti-aliasing filter can be supplemented by

1728-475: A limited and variable lifespan due to both inherent and manufacturing quality issues. With vinyl records, there will be some loss in fidelity on each playing of the disc. This is due to the wear of the stylus in contact with the record surface. Magnetic tapes, both analog and digital, wear from friction between the tape and the heads, guides, and other parts of the tape transport as the tape slides over them. The brown residue deposited on swabs during cleaning of

1824-411: A modern digital system may compress input signals so that digital full-scale cannot be reached Unlike analog duplication, digital copies are exact replicas that can be duplicated indefinitely and without generation loss , in principle. Error correction allows digital formats to tolerate significant media deterioration though digital media is not immune to data loss. Consumer CD-R compact discs have

1920-454: A new type of AIFF which is, in effect, an alternative little-endian byte order format. Because the AIFF architecture has no provision for alternative byte order, Apple used the existing AIFF-C compression architecture, and created a "pseudo-compressed" codec called sowt ( twos spelled backwards). The only difference between a standard AIFF file and an AIFF-C/sowt file is the byte order; there

2016-412: A piece of audio equipment adds to the original signal can be quantified. Mathematically, this can be expressed by means of the signal-to-noise ratio (SNR or S/N ratio). Sometimes the maximum possible dynamic range of the system is quoted instead. With digital systems, the quality of reproduction depends on the analog-to-digital and digital-to-analog conversion steps, and does not depend on the quality of

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2112-522: 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

2208-537: A recordable optical technology which markets itself as remaining readable for 1,000 years, is available in certain markets, but as of late 2020 has never been sold in the CD-R format. (Sound could, however, be stored on an M-DISC DVD-R using the DVD-Audio format.) For electronic audio signals, sources of noise include mechanical, electrical and thermal noise in the recording and playback cycle. The amount of noise that

2304-580: 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

2400-402: A response extending up to 15 kHz at full (0 dB) recording level. At lower levels (−10 dB), cassettes are typically limited to 20 kHz due to self-erasure of the tape media. The frequency response for a conventional LP player might be 20 Hz to 20 kHz, ±3 dB. The low-frequency response of vinyl records is restricted by rumble noise (described above), as well as

2496-402: A sample is determined by the number of binary digits used. This is called the resolution, and is usually referred to as the bit depth in the context of PCM audio. The quantization noise level is directly determined by this number, decreasing exponentially (linearly in dB units) as the resolution increases. With an adequate bit depth, random noise from other sources will dominate and completely mask

2592-446: A signal with a filter, the output signal may differ in time from the signal at the input, which is measured as its phase response . All analog equalizers exhibit this behavior, with the amount of phase shift differing in some pattern, and centered around the band that is being adjusted. Although this effect alters the signal in a way other than a strict change in frequency response, it is usually not objectionable to listeners. Because

2688-521: 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

2784-442: 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

2880-503: 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. Comparison of analog and digital recording Sound can be recorded and stored and played using either digital or analog techniques. Both techniques introduce errors and distortions in

2976-404: A tape machine's tape path is actually particles of magnetic coating shed from tapes. Sticky-shed syndrome is a prevalent problem with older tapes. Tapes can also suffer creasing, stretching, and frilling of the edges of the plastic tape base, particularly from low-quality or out-of-alignment tape decks. When a CD is played, there is no physical contact involved as the data is read optically using

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3072-429: A well-designed 16- bit channel." DVD-Audio and most modern professional recording equipment allows for samples of 24 bits. Analog systems do not necessarily have discrete digital levels in which the signal is encoded. Consequently, the accuracy to which the original signal can be preserved is instead limited by the intrinsic noise-floor and maximum signal level of the media and the playback equipment. Since analog media

3168-771: Is a form of noise characteristic caused by imperfections in the bearings of turntables. The platter tends to have a slight amount of motion besides the desired rotation and the turntable surface also moves up, down and side-to-side slightly. This additional motion is added to the desired signal as noise, usually of very low frequencies, creating a rumbling sound during quiet passages. Very inexpensive turntables sometimes used ball bearings , which are very likely to generate audible amounts of rumble. More expensive turntables tend to use massive sleeve bearings , which are much less likely to generate offensive amounts of rumble. Increased turntable mass also tends to lead to reduced rumble. A good turntable should have rumble at least 60 dB below

3264-451: Is a listening test, where the audio component is simply used in the context for which it was designed. This test is popular with hi-fi reviewers, where the component is used for a length of time by the reviewer who then will describe the performance in subjective terms. Common descriptions include whether the component has a bright or warm sound, or how well the component manages to present a spatial image . Another type of subjective test

3360-402: Is captured above what is generally considered to be the human hearing frequency range . The higher sample rates impose less restrictions on anti-aliasing filter implementation which can result in both lower complexity and less signal distortion. Work done in 1981 by Muraoka et al. showed that music signals with frequency components above 20 kHz were only distinguished from those without by

3456-415: Is composed of molecules , the smallest microscopic structure represents the smallest quantization unit of the recorded signal. Natural dithering processes, like random thermal movements of molecules, the nonzero size of the reading instrument, and other averaging effects, make the practical limit larger than that of the smallest molecular structural feature. A theoretical LP composed of perfect diamond, with

3552-439: Is dependent on the physical and electronic capabilities of the analog circuits. The S/N ratio of a digital system may be limited by the bit depth of the digitization process, but the electronic implementation of conversion circuits introduces additional noise. In an analog system, other natural analog noise sources exist, such as flicker noise and imperfections in the recording medium. Other performance differences are specific to

3648-483: Is digitally sampled using native methods (without dither), the amplitude of the audio signal will simply be rounded to the nearest representation. This process is called quantization, and these small errors in the measurements are manifested aurally as low level noise or distortion. This form of distortion, sometimes called granular or quantization distortion, has been pointed to as a fault of some digital systems and recordings particularly some early digital recordings, where

3744-422: Is done under more controlled conditions and attempts to remove possible bias from listening tests. These sorts of tests are done with the component hidden from the listener, and are called blind tests . To prevent possible bias from the person running the test, the blind test may be done so that this person is also unaware of the component under test. This type of test is called a double-blind test. This sort of test

3840-476: Is mathematically sufficient to capture all the information contained in a signal having frequency components less than or equal to 20 kHz. The sampling theorem also requires that frequency content above the Nyquist frequency be removed from the signal before sampling it. This is accomplished using anti-aliasing filters which require a transition band to sufficiently reduce aliasing. The bandwidth provided by

3936-408: Is no compression involved at all. Apple uses this new little-endian AIFF type as its standard on macOS. When a file is imported to or exported from iTunes in "AIFF" format, it is actually AIFF-C/sowt that is being used. When audio from an audio CD is imported by dragging to the macOS Desktop, the resulting file is also an AIFF-C/sowt. In all cases, Apple refers to the files simply as "AIFF", and uses

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4032-487: Is not clear, however, if detection thresholds obtained in the present study would really represent the limit of auditory resolution or it would be limited by resolution of equipment. Distortions due to very small jitter may be smaller than distortions due to non-linear characteristics of loudspeakers. Ashihara and Kiryu [8] evaluated linearity of loudspeaker and headphones. According to their observation, headphones seem to be more preferable to produce sufficient sound pressure at

4128-406: Is often used to evaluate the performance of lossy audio compression . Critics of double-blind tests see them as not allowing the listener to feel fully relaxed when evaluating the system component, and can therefore not judge differences between different components as well as in sighted (non-blind) tests. Those who employ the double-blind testing method may try to reduce listener stress by allowing

4224-439: 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

4320-431: Is the form of high-level signal error. Some early analog-to-digital converters displayed non-benign behaviour when in overload, where the overloading signals were 'wrapped' from positive to negative full-scale. Modern converter designs based on sigma-delta modulation may become unstable in overload conditions. It is usually a design goal of digital systems to limit high-level signals to prevent overload. To prevent overload,

4416-456: Is the possibility that such signals could push the system into overload. With high level signals, analog magnetic tape approaches saturation , and high frequency response drops in proportion to low frequency response. While undesirable, the audible effect of this can be reasonably unobjectionable. In contrast, digital PCM recorders show non-benign behaviour in overload; samples that exceed the peak quantization level are simply truncated, clipping

4512-469: 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),

4608-643: The .aiff (or .aif ) or .caf extension regardless of type. An AIFF file is divided into a number of chunks. Each chunk is identified by a chunk ID more broadly referred to as FourCC . Types of chunks found in AIFF files: AIFF files can store metadata in Name, Author, Comment, Annotation, and Copyright chunks. An ID3v2 tag chunk can also be embedded in AIFF files, as well as an Application Chunk with Extensible Metadata Platform (XMP) data in it. AIFF supports only uncompressed PCM data. AIFF-C also supports compressed audio formats, which can be specified in

4704-577: The .aiff extension. For the vast majority of users this technical situation is completely unnoticeable and irrelevant. The sound quality of standard AIFF and AIFF-C/sowt are identical, and the data can be converted back and forth without loss. Users of older audio applications, however, may find that an AIFF-C/sowt file will not play, or will prompt the user to convert the format on opening, or will play as static. All traditional AIFF and AIFF-C files continue to work normally on macOS, and many third-party audio applications as well as hardware continue to use

4800-522: The 44,100 Hz sampling frequency used by the standard for audio CDs is sufficiently wide to cover the entire human hearing range , which roughly extends from 20 Hz to 20 kHz. Professional digital recorders may record higher frequencies, while some consumer and telecommunications systems record a more restricted frequency range. Some analog tape manufacturers specify frequency responses up to 20 kHz, but these measurements may have been made at lower signal levels. Compact Cassettes may have

4896-552: 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

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4992-720: 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

5088-458: The least significant bit of the digital system. Dither algorithms also commonly have an option to employ some kind of noise shaping , which pushes the frequency of much of the dither noise to areas that are less audible to human ears, lowering the level of the noise floor apparent to the listener. Dither is commonly applied during mastering before final bit depth reduction, and also at various stages of DSP . One aspect that may degrade

5184-416: The "COMM" chunk. The compression type is "NONE" for PCM audio data. The compression type is accompanied by a printable name. Common compression types and names include, but are not limited to: Digital audio 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

5280-432: The 16 bits of audio CD. Meridian Audio founder John Robert Stuart stresses that with the correct dither , the resolution of a digital system is theoretically infinite, and that it is possible, for example, to resolve sounds at −110 dB (below digital full-scale) in a well-designed 16-bit channel. There are some differences in the behaviour of analog and digital systems when high level signals are present, where there

5376-434: 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

5472-430: The 1990s and 2000s. 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

5568-457: The Science , subjects using the same high-sample-rate reproduction equipment could not discern any audible difference between program material identically filtered to remove frequencies above 20 kHz versus 40 kHz. This demonstrates that presence or absence of ultrasonic content does not explain aural variation between sample rates. He posits that variation is due largely to performance of

5664-412: The ability to remove noise from the algorithms and modifications to make the parameters more flexible. On the other hand, other engineers also feel that the modeling is still inferior to the genuine outboard components and still prefer to mix "outside the box". Subjective evaluation attempts to measure how well an audio component performs according to the human ear. The most common form of subjective test

5760-420: The audible range, producing a kind of distortion called aliasing . Aliasing is prevented in digital systems by an anti-aliasing filter . However, designing an analog filter that precisely removes all frequency content exactly above or below a certain cutoff frequency, is impractical. Instead, a sample rate is usually chosen which is above the Nyquist requirement. This solution is called oversampling , and allows

5856-442: 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

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5952-419: The band-limiting filters in converters. These results suggest that the main benefit to using higher sample rates is that it pushes consequential phase distortion from the band-limiting filters out of the audible range and that, under ideal conditions, higher sample rates may not be necessary. Dunn (1998) examined the performance of digital converters to see if these differences in performance could be explained by

6048-426: The band-limiting filters used in converters and looking for the artifacts they introduce. A signal is recorded digitally by an analog-to-digital converter , which measures the amplitude of an analog signal at regular intervals specified by the sampling rate, and then stores these sampled numbers in computer hardware. Numbers on computers represent a finite set of discrete values, which means that if an analog signal

6144-466: 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

6240-405: The behavior exhibited by the systems due to these methods. The dynamic range of digital audio systems can exceed that of analog audio systems. Consumer analog cassette tapes have a dynamic range of 60 to 70 dB. Analog FM broadcasts rarely have a dynamic range exceeding 50 dB. The dynamic range of a direct-cut vinyl record may surpass 70 dB. Analog studio master tapes can have

6336-482: 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

6432-535: 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

6528-576: The design considerations associated with them. Instead, the limits of analog storage formats are determined by the physical properties of their construction. CD quality audio is sampled at 44,100 Hz ( Nyquist frequency = 22.05 kHz) and at 16 bits. Sampling the waveform at higher frequencies and allowing for a greater number of bits per sample allows noise and distortion to be reduced further. DAT can sample audio at up to 48 kHz, while DVD-Audio can be 96 or 192 kHz and up to 24 bits resolution. With any of these sampling rates, signal information

6624-399: 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

6720-408: The digital release was said to be inferior to the analog version. However, "if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white, uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel." The range of possible values that can be represented numerically by

6816-536: The digital stream affects a subsequent part as it flows through the system, and power supply induced jitter, where noise from the power supply causes irregularities in the timing of signals in the circuits it powers. The accuracy of a digital system is dependent on the sampled amplitude values, but it is also dependent on the temporal regularity of these values. The analog versions of this temporal dependence are known as pitch error and wow-and-flutter. Periodic jitter produces modulation noise and can be thought of as being

6912-640: The ear drums with smaller distortions than loudspeakers. After initial recording, it is common for the audio signal to be altered in some way, such as with the use of compression , equalization , delays and reverb . With analog, this comes in the form of outboard hardware components , and with digital, the same is typically accomplished with plug-ins in a digital audio workstation (DAW). A comparison of analog and digital filtering shows technical advantages to both methods. Digital filters are more precise and flexible. Analog filters are simpler, can be more efficient and do not introduce latency. When altering

7008-570: 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

7104-429: The equivalent of analog flutter. Random jitter alters the noise floor of the digital system. The sensitivity of the converter to jitter depends on the design of the converter. It has been shown that a random jitter of 5  ns may be significant for 16 bit digital systems. In 1998, Benjamin and Gannon researched the audibility of jitter using listening tests. They found that the lowest level of jitter to be audible

7200-399: 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

7296-433: The high levels of performance possible with digital audio, including excellent linearity in the audible band and low levels of noise and distortion. Two prominent differences in performance between the two methods are the bandwidth and the signal-to-noise ratio (S/N ratio). The bandwidth of the digital system is determined, according to the Nyquist frequency , by the sample rate used. The bandwidth of an analog system

7392-450: The musical note of a sample , for use by hardware samplers and musical applications. The file extension for the standard AIFF format is .aiff or .aif . For the compressed variants it is supposed to be .aifc , but .aiff or .aif are accepted as well by audio applications supporting the format. With the development of the OS X operating system now known as macOS , Apple created

7488-578: 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

7584-431: The performance of a digital system is jitter . This is the phenomenon of variations in time from what should be the correct spacing of discrete samples according to the sample rate. This can be due to timing inaccuracies of the digital clock. Ideally, a digital clock should produce a timing pulse at exactly regular intervals. Other sources of jitter within digital electronic circuits are data-induced jitter, where one part of

7680-521: The physical and electrical characteristics of the entire pickup arm and transducer assembly. The high-frequency response of vinyl depends on the cartridge. CD4 records contained frequencies up to 50 kHz. Frequencies of up to 122 kHz have been experimentally cut on LP records. Digital systems require that all high-frequency signal content above the Nyquist frequency must be removed prior to sampling, which, if not done, will result in these ultrasonic frequencies "folding over" into frequencies in

7776-468: The quality of the turntable will have a large effect on the level of wow and flutter. A good turntable will have wow and flutter values of less than 0.05%, which is the speed variation from the mean value. Wow and flutter can also be present in the recording, as a result of the imperfect operation of the recorder. Owing to their use of precision crystal oscillators for their timebase , digital systems are not subject to wow and flutter. For digital systems,

7872-400: The quantization noise. The Redbook CD standard uses 16 bits, which keeps the quantization noise 96 dB below maximum amplitude, far below a discernible level with almost any source material. The addition of effective dither means that, "in practical terms, the resolution is limited by our ability to resolve sounds in noise. ... We have no problem measuring (and hearing) signals of –110dB in

7968-467: The recording medium, provided it is adequate to retain the digital values without error. Digital media capable of bit-perfect storage and retrieval have been commonplace for some time, since they were generally developed for software storage which has no tolerance for error. The process of analog-to-digital conversion will, according to theory, always introduce quantization distortion. This distortion can be rendered as uncorrelated quantization noise through

8064-439: 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

8160-426: 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

8256-401: The specified output level from the pick-up. Because they have no moving parts in the signal path, digital systems are not subject to rumble. Wow and flutter are a change in frequency of an analog device and are the result of mechanical imperfections. Wow is a form of flutter that occurs at a slower rate. Wow and flutter are most noticeable on signals which contain pure tones. For LP records,

8352-404: The standard AIFF big-endian byte order. Apple has also created another recent extension to the AIFF format in the form of Apple Loops used by GarageBand and Logic Pro , which allows the inclusion of data for pitch and tempo shifting by an application in the more common variety, and MIDI -sequence data and references to GarageBand playback instruments in another variety. Apple Loops use either

8448-400: 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

8544-426: The systems under comparison, such as the ability for more transparent filtering algorithms in digital systems and the harmonic saturation and speed variations of analog systems. The dynamic range of an audio system is a measure of the difference between the smallest and largest amplitude values that can be represented in a medium. Digital and analog differ in both the methods of transfer and storage, as well as

8640-418: The upper limit of the frequency response is determined by the sampling frequency . The choice of sample sampling frequency in a digital system is based on the Nyquist–Shannon sampling theorem . This states that a sampled signal can be reproduced exactly as long as it is sampled at a frequency greater than twice the bandwidth of the signal, the Nyquist frequency . Therefore, a sampling frequency of 40 kHz

8736-509: The use of dither . The magnitude of this noise or distortion is determined by the number of quantization levels. In binary systems this is determined by and typically stated in terms of the number of bits . Each additional bit adds approximately 6 dB in possible SNR (e.g. 24 x 6 = 144 dB for 24-bit and 120 dB for 20-bit quantization). The 16-bit digital system of Red Book audio CD has 2 = 65,536 possible signal amplitudes, theoretically allowing for an SNR of 98  dB . Rumble

8832-403: The variables involved can be precisely specified in the calculations, digital filters can be made to objectively perform better than analog components. Other processing such as delay and mixing can be done exactly. Digital filters are also more versatile. For example, the linear phase equalizer does not introduce frequency-dependent phase shift. This filter may be implemented digitally using

8928-453: The waveform squarely, which introduces distortion in the form of large quantities of higher-frequency harmonics. In principle, PCM digital systems have the lowest level of nonlinear distortion at full signal amplitude. The opposite is usually true of analog systems, where distortion tends to increase at high signal levels. A study by Manson (1980) considered the requirements of a digital audio system for high quality broadcasting. It concluded that

9024-511: Was around 10 ns ( rms ). This was on a 17 kHz sine wave test signal. With music, no listeners found jitter audible at levels lower than 20 ns. A paper by Ashihara et al. (2005) attempted to determine the detection thresholds for random jitter in music signals. Their method involved ABX listening tests . When discussing their results, the authors commented that: So far, actual jitter in consumer products seems to be too small to be detected at least for reproduction of music signals. It

9120-446: Was noticed that CD was far more revealing of the acoustics and ambient background noise of the recording environment. For this reason, recording techniques developed for analog disc, e.g., microphone placement, needed to be adapted to suit the new digital format. Some analog recordings were remastered for digital formats. Analog recordings made in natural concert hall acoustics tended to benefit from remastering. The remastering process

9216-499: 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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