View source for Comparison of analog and digital recording

{{Short description|Audio performance differences between technologies}}
{{Use American English|date=July 2020}}
{{Use dmy dates|date=July 2020}}

[[Sound]] can be [[audio recording|recorded]] and stored and played using either [[Digital signal (signal processing)|digital]] or [[analog signal|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]].<ref name="omni_con">{{cite magazine |last=Liversidge |first=Anthony |url=http://nest.machinecode.org/intricate/omni_magazine/Omni%20v17%20%235%20February%201995.html |title=Analog versus digital: has vinyl been wrongly dethroned by the music industry? |magazine=Omni |date=February 1995 |volume=17 |issue=5 |access-date=22 March 2017 |archive-date=7 March 2015 |archive-url=https://web.archive.org/web/20150307064638/http://nest.machinecode.org/intricate/omni_magazine/Omni%20v17%20%235%20February%201995.html |url-status=dead }}</ref> 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.<ref name="SonyServiceCentre"/>{{rp|7}}

Two prominent differences in performance between the two methods are the [[Bandwidth (signal processing)|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 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 [[Audio bit depth|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 the systems under comparison, such as the ability for more transparent [[digital filter|filtering algorithms]] in digital systems<ref name="dspguide21">{{cite web |url=http://www.dspguide.com/ch21/1.html|title=Chapter 21: Filter Comparison|publisher=dspguide.com|access-date=2012-09-13}}</ref> and the [[distortion (music)|harmonic saturation]] and [[Wow (recording)|speed variations]] of analog systems.

==Dynamic range==
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 the behavior exhibited by the systems due to these methods.

The dynamic range capability of digital audio systems far exceeds that of analog audio systems. Consumer analog [[compact audio cassette|cassette]] tapes have a dynamic range of between 50 and 75&nbsp;dB. Analog [[FM broadcast]]s rarely have a dynamic range exceeding 50&nbsp;dB.<ref>{{cite web |author=Garrison |first=Mark |date=23 Sep 2011 |title=Encyclopedia of Home Recording: Dynamic Range |work=Music & Audio Envato Tuts+ |url=https://music.tutsplus.com/articles/encyclopedia-of-home-recording-dynamic-range--audio-11948}}</ref> Analog studio master tapes can have a dynamic range of up to 77&nbsp;dB.<ref>{{cite web |url=http://www.theaudioarchive.com/TAA_Tape_Studer_A820.htm |title=State-of-the-Art Audio Transfer |publisher=The Audio Archive |access-date=2018-05-14 |quote=Signal-to-Noise NAB (1/4-inch two-track 2.0 mm track, RMS, A-weighted) 30 ips - 75 dB}}</ref> An LP made out of perfect vinyl would have a theoretical dynamic range of 70&nbsp;dB,<ref name="Lesurf">{{cite web |author=Lesurf |first=Jim |date=May 18, 2000 |title=The 'digital' defects of the long-playing record |url=https://www.st-andrews.ac.uk/~www_pa/Scots_Guide/iandm/part12/page2.html |access-date=September 22, 2017 |publisher=[[University of St. Andrews]]}}</ref> though measurements indicate actual performance in the 60 to 70&nbsp;dB range.<ref>{{cite magazine |author=Fremer |first=Michael |date=January 6, 1999 |title=Pass Aleph Ono phono preamplifier |url=https://www.stereophile.com/phonopreamps/199pass/index1.html |access-date=2018-05-14 |magazine=[[Stereophile]]}}</ref>{{fv|reason=This article is actually arguing for a higher dynamic range than the commonly accepted 60-70 dB figure. It proposes measuring the ratio between loudest playback and the noise from an idle preamp. Not a valid proposal IMO; the needle should be on the record and the vinyl should be moving. In any case, although it is (disparagingly) mentioned here, a better source for the 60-70 dB figure is needed.|date=December 2024}}

Compare this to digital recording. Typically, a 16-bit digital recording has a dynamic range of between 90 and 95&nbsp;dB.<ref name="Metzler2005">{{cite book |last=Metzler |first=Bob |date=2005 |title=The Audio Measurement Handbook |edition=2 |publisher=Audio Precision, USA |url=https://www.ap.com/download/the-audio-measurement-handbook-2/?wpdmdl=5931 |access-date=2008-03-09 |url-access=registration}}</ref>{{rp|132}}

The benefits of using digital recorders with greater than 16-bit accuracy can be applied to 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&nbsp;dB (below digital full-scale) in a well-designed 16-bit channel.<ref name="Stuart"/>{{rp|3}}

===Overload conditions===
There are some differences in the behaviour of analog and digital systems when high level signals are present, where there is the possibility that such signals could push the system into overload. With high level signals, analog magnetic tape approaches [[saturation (magnetic)|saturation]], and high frequency response drops in proportion to low frequency response. While undesirable, the audible effect of this can be reasonably unobjectionable.<ref name="Elsea1996">{{cite web |last=Elsea |first=Peter |date=1996 |url=http://arts.ucsc.edu/ems/music/tech_background/TE-19/teces_19.html |title=Analog Recording of Sound |publisher=Electronic Music Studios at the University of California, Santa Cruz |access-date=2008-03-09 |archive-url=https://web.archive.org/web/20091016023237/http://arts.ucsc.edu/ems/Music/tech_background/TE-19/teces_19.html |archive-date=16 October 2009 |url-status=dead }}</ref> In contrast, digital PCM recorders show non-benign behaviour in overload;<ref name="Dunn2003"/>{{rp|65}} samples that exceed the peak quantization level are simply truncated, clipping 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 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 clipping|soft clipper]]' be used to prevent the system from becoming overloaded.<ref name="Manson1980">{{cite web |last=Manson |first=W. |date=1980 |url=http://www.bbc.co.uk/rd/pubs/reports/1980-15.pdf |title=Digital Sound: studio signal coding resolution for broadcasting |publisher=BBC Research Department, Engineering Division |page=8}}</ref>

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 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.<ref name="Dunn2003"/>{{rp|65}} To prevent overload, a modern digital system may [[dynamic range compression|compress]] input signals so that digital full-scale cannot be reached<ref name="Jones2003">{{cite conference |last1=Jones |first1=Wayne |last2=Wolfe |first2=Michael |last3=Tanner |first3=Theodore C. Jr. |last4=Dinu |first4=Daniel |date=March 2003 |url=http://ap.com/library/whitepapers.htm |archive-url=https://web.archive.org/web/20080307052354/http://ap.com/library/whitepapers.htm  |archive-date=2008-03-07 |title=Testing Challenges in Personal Computer Audio Devices |conference=114th AES Convention |access-date=2008-03-09}}</ref>{{rp|4}}

==Physical degradation==
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 a limited and variable lifespan due to both inherent and manufacturing quality issues.<ref>{{cite web | work=myce.com| url=http://www.myce.com/news/cd-recordable-discs-unreadable-in-less-than-two-years-6450 | title=CD-R Unreadable in Less Than Two Years| date=19 August 2003 | access-date=2007-02-01}}</ref>

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 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 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.<ref>{{cite web |url=http://www.clir.org/pubs/reports/pub121/pub121.pdf |title=Care and Handling of CDs and DVDs |first=Fred R |last=Byers |publisher=Council on Library and Information Resources |date=October 2003 |access-date=27 July 2014}}</ref> [[M-DISC]], 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 recordable|DVD-R]] using the [[DVD-Audio]] format.)

==Noise==
For electronic audio signals, sources of noise include mechanical, electrical and thermal noise in the recording and playback cycle. The amount of noise that 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 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 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 [[Audio bit depth|number of bits]]. Each additional bit adds approximately 6&nbsp;dB in possible SNR (e.g. 24 x 6 = 144&nbsp;dB for 24-bit and 120&nbsp;dB for 20-bit quantization). The 16-bit digital system of [[Red Book (audio CD standard)|Red Book audio CD]] has 2<sup>16</sup> = 65,536 possible signal amplitudes, theoretically allowing for an SNR of 98&nbsp;[[Decibel|dB]].<ref name="SonyServiceCentre">{{cite book
|editor1-first=Jan |editor1-last=Maes
|editor2-first=Marc|editor2-last=Vercammen
|date=2001
|edition=4
|title=Digital Audio Technology: A guide to CD, MiniDisc, SACD, DVD(A), MP3 and DAT
|publisher=Focal Press
|isbn=0240516540
|quote=A 16-bit system, therefore, gives a theoretical signal-to-noise ratio of 98 dB...
}}</ref>{{rp|49}}

===Rumble===
Rumble is a form of noise characteristic caused by imperfections in the [[Bearing (mechanical)|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 bearing]]s, which are very likely to generate audible amounts of rumble. More expensive turntables tend to use massive [[sleeve bearing]]s, 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&nbsp;dB below the specified output level from the pick-up.<ref name="Driscoll 1980">Driscoll, R. (1980). ''Practical Hi-Fi Sound'', 'Analogue and digital', pages 61–64; 'The pick-up, arm and turntable', pages 79–82. Hamlyn. {{ISBN|0-600-34627-7}}.</ref>{{rp|79–82}} Because they have no moving parts in the signal path, digital systems are not subject to rumble.

===Wow and flutter===
[[wow (recording)|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, 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.<ref name="Driscoll 1980"/> 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 oscillator]]s for their [[timebase]], digital systems are not subject to wow and flutter.

==Frequency response==
For digital systems, 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 (signal processing)|bandwidth]] of the signal, the [[Nyquist frequency]]. Therefore, a sampling frequency of 40&nbsp;kHz is mathematically sufficient to capture all the information contained in a signal having frequency components less than or equal to 20&nbsp;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 filter]]s which require a [[transition band]] to sufficiently reduce aliasing. The bandwidth provided by the [[44,100 Hz]] sampling frequency used by the [[Red Book (CD standard)|standard for audio CDs]] is sufficiently wide to cover the entire human [[hearing range]], which roughly extends from 20&nbsp;Hz to 20&nbsp;kHz.<ref name="SonyServiceCentre"/>{{rp|108}} 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&nbsp;kHz, but these measurements may have been made at lower signal levels.<ref name="Driscoll 1980"/> [[Compact Cassette]]s may have a response extending up to 15&nbsp;kHz at full (0&nbsp;dB) recording level.<ref name="Stark1989">{{cite book |last=Stark |first=C. |date=1989 |publisher=[[Encyclopædia Britannica]] |edition=15 |volume=27 |title=Macropaedia article 'Sound' |section=High-fidelity concepts and systems |page=625}}</ref>  At lower levels (−10&nbsp;dB), cassettes are typically limited to 20&nbsp;kHz due to [[self-erasure]] of the tape media.

The frequency response for a conventional LP player might be 20&nbsp;Hz to 20&nbsp;kHz, ±3&nbsp;dB. The low-frequency response of vinyl records is restricted by rumble noise (described above), as well as the physical and electrical characteristics of the entire pickup arm and transducer assembly. The high-frequency response of vinyl depends on the cartridge. [[Quadraphonic|CD4]] records contained frequencies up to 50&nbsp;kHz. Frequencies of up to 122&nbsp;kHz have been experimentally cut on LP records.<ref>{{cite web |url=http://www.positive-feedback.com/Issue2/mastering.htm |title=mastering |publisher=Positive-feedback.com |access-date=2012-08-15}}</ref>

===Aliasing===
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 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.<ref name="uadt14">Thompson, Dan. ''Understanding Audio''. Berklee Press, 2005, ch. 14.</ref> Instead, a sample rate is usually chosen which is above the Nyquist requirement. This solution is called [[oversampling]], and allows 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 (electrical)|ripple]], temperature dependence of filters etc.<ref name="Hawksford1991">{{cite conference |last=Hawksford |first=Malcolm |archive-date=2007-09-29 |archive-url=https://web.archive.org/web/20070929134414/http://www.essex.ac.uk/ese/research/audio_lab/malcolmspubdocs/C27%20AES%20lecture%20Introduction%20to%20digital%20audio.pdf |url=http://www.essex.ac.uk/ese/research/audio_lab/malcolmspubdocs/C27%20AES%20lecture%20Introduction%20to%20digital%20audio.pdf |title=Introduction to Digital Audio Images of Audio |conference=Proceedings of the 10th International AES Conference |location=London |date=September 1991 |access-date=2008-03-09}}</ref>{{rp|8}} Using an oversampling design and [[delta-sigma modulation]], a less aggressive analog anti-aliasing filter can be supplemented by a digital filter.<ref name="uadt14"/> 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.<ref name="Hawksford1991"/>{{rp|18}} However, the digital anti-aliasing filter may introduce degradations due to time domain response particularly at lower sample rates.<ref>{{cite web|last1=Story|first1=Mike|title=A Suggested Explanation For (Some Of) The Audible Differences Between High Sample Rate And Conventional Sample Rate Audio Material |date=September 1997|url=http://sdg-master.com:80/lesestoff/aes97ny.pdf |publisher=dCS Ltd|archive-date=28 November 2009|archive-url=https://web.archive.org/web/20091128021651/http://sdg-master.com:80/lesestoff/aes97ny.pdf|url-status=live}}</ref><ref>{{cite web|last1=Robjohns|first1=Hugh|title= MQA Time-domain Accuracy & Digital Audio Quality|url= https://www.soundonsound.com/techniques/mqa-time-domain-accuracy-digital-audio-quality|website=soundonsound.com|date=August 2016|publisher= Sound On Sound|archive-date=10 March 2023|archive-url= https://web.archive.org/web/20230310175409/https://www.soundonsound.com/techniques/mqa-time-domain-accuracy-digital-audio-quality|url-status=live}}</ref>

Analog systems are not subject to a Nyquist limit or aliasing and thus do not require anti-aliasing filters or any of the design considerations associated with them. Instead, the limits of analog storage formats are determined by the physical properties of their construction.

==Sampling rates==
CD quality audio is sampled at [[44,100 Hz]] ([[Nyquist frequency]] = 22.05&nbsp;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&nbsp;kHz, while [[DVD-Audio]] can be 96 or 192&nbsp;kHz and up to 24 bits resolution. With any of these sampling rates, signal information is captured above what is generally considered to be the [[human hearing range|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.<ref>{{cite journal |last1=Muraoka |first1=Teruo |last2=Iwahara |first2=Makoto |last3=Yamada |first3=Yasuhiro |year=1981 |title=Examination of Audio-Bandwidth Requirements for Optimum Sound Signal Transmission |journal=Journal of the Audio Engineering Society |volume=29 |issue=1/2 |pages=2–9}}</ref> showed that music signals with frequency components above 20&nbsp;kHz were only distinguished from those without by a few of the 176 test subjects.<ref name="Kaoru2001">{{cite conference |last1=Kaoru |first1=A. |last2=Shogo |first2=K |url=http://www.aes.org/e-lib/browse.cfm?elib=10005 |title=Detection threshold for tones above 22 kHz |publisher=[[Audio Engineering Society]] |postscript=Paper 5401 |conference=110th AEC Convention |date=2001}}</ref> 
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."<ref>{{Cite report|last1=Nishiguchi |first1=Toshiyuki |last2=Iwaki |first2=Masakazu |last3=Ando |first3=Akio |year=2004 |title=Perceptual Discrimination between Musical Sounds with and without Very High Frequency Components |work=NHK Laboratories Note No. 486 |url=http://www.nhk.or.jp/strl/publica/labnote/lab486.html |publisher=[[NHK]] |access-date=August 15, 2012 |url-status=dead |archive-url=https://web.archive.org/web/20151016212854/http://www.nhk.or.jp/strl/publica/labnote/lab486.html |archive-date=October 16, 2015 }}</ref>

In blind listening tests conducted by [[Bob Katz]] in 1996, recounted in his book ''Mastering Audio: The Art and 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&nbsp;kHz versus 40&nbsp;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 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.<ref name="masteringkatz">{{cite book |last=Katz |first=Bob |date=2015 |title=Mastering Audio: The Art and the Science |publisher=Focal Press |isbn=978-0240818962 |edition=3rd |page=316-318}}</ref> Dunn (1998) examined the performance of digital converters to see if these differences in performance could be explained by the band-limiting filters used in converters and looking for the artifacts they introduce.<ref name="Dunn1998">{{cite web |url=http://www.nanophon.com/audio/antialia.pdf |title=Anti-alias and anti-image filtering: The benefits of 96kHz sampling rate formats for those who cannot hear above 20kHz |publisher=Nanophon Limited |date=1998 |first=Julian |last=Dunn |access-date=27 July 2014}}</ref>

==Quantization==
[[File:Pcm.svg|thumb|An illustration of quantization of a sampled audio waveform using 4 bits]]

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. [[Computer number format|Numbers on computers]] represent a finite set of discrete values, which means that if an analog signal 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 the digital release was said to be inferior to the analog version.<ref name="Knee1995">{{cite conference |last1=Knee |first1=Anthony B. |last2=Hawksford |first2=Malcolm J. |url=http://www.aes.org/e-lib/browse.cfm?elib=7763 |title=Evaluation of Digital Systems and Digital Recording Using Real Time Audio Data |conference=98th AES Convention |date=February 1995 |pages=3}}</ref> 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."<ref name="Stuart"/>{{rp|6}}

The range of possible values that can be represented numerically by 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 the quantization noise.  The Redbook CD standard uses 16 bits, which keeps the quantization noise 96&nbsp;dB below maximum amplitude, far below a discernible level with almost any source material.<ref>{{cite web |first=Jeffrey |last=Hass |title=Chapter 5: Principles of Digital Audio |date=2013 |website=Center for Electronic and Computer Music |publisher=Indiana University |url=http://www.indiana.edu/~emusic/etext/digital_audio/chapter5_quantize.shtml}}</ref> 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 a well-designed 16-
bit channel."<ref name="Stuart"/> 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.

===Quantization in analog media===
Since analog media is composed of [[molecule]]s, the smallest [[Solid#Microscopic description|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 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.<ref name="Lesurf" />

===Dither as a solution===
[[File:Dithering example dithered 16color.png|thumb|alt=An illustration of dither used in image processing.|An illustration of dither used in image processing. A random deviation has been inserted before reducing the palette to only 16 colors, which is analogous to the effect of dither on an audio signal.]]

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,<ref name="Dunn2003">{{cite web |last=Dunn |first=Julian |date=2003 |url=http://www.ap.com/library/technotes.htm |archive-url=https://web.archive.org/web/20070320190428/http://www.ap.com/library/technotes.htm |archive-date=2007-03-20 |title=Measurement Techniques for Digital Audio: Audio Precision Application Note #5 |publisher=Audio Precision, Inc. |access-date=2008-03-09}}</ref>{{rp|143}} and allows signal information to be retained below the [[least significant bit]] of the digital system.<ref name="Stuart">{{cite web |last=Stuart |first=J. |url=http://www.meridian-audio.com/ara/coding2.pdf |title=Coding High Quality Digital Audio |publisher=Meridian Audio Ltd |access-date=2008-03-09 |archive-url=https://web.archive.org/web/20071011100455/http://www.meridian-audio.com/ara/coding2.pdf |archive-date=2007-10-11 |url-status=dead }} This article is substantially the same as Stuart's 2004 JAES article [http://www.aes.org/e-lib/browse.cfm?elib=12986 "Coding for High-Resolution Audio Systems"], ''Journal of the Audio Engineering Society'', Volume 52 Issue 3 pp.&nbsp;117–144; March 2004.</ref>{{rp|3}}

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 [[audio mastering|mastering]] before final bit depth reduction,<ref name="masteringkatz"/> and also at various stages of [[digital signal processing|DSP]].

==Timing jitter==
One aspect that may degrade 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 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 the equivalent of analog flutter.<ref name="Watkinson1995">{{cite book |last1=Rumsey |first1=F. |last2=Watkinson |first2=J |date=1995 |title=The Digital Interface Handbook |edition=2 |chapter=Sections 2.5 and 6 |pages=37, 154–160 |publisher=Focal Press}}</ref> Random jitter alters the noise floor of the digital system. The sensitivity of the converter to jitter depends on the design of the converter.<ref name="Dunn2003"/> It has been shown that a random jitter of 5&nbsp;[[nanosecond|ns]] may be significant for 16 bit digital systems.<ref name="Watkinson1995"/>

In 1998, Benjamin and Gannon researched the audibility of jitter using listening tests.<ref name="Dunn2003"/>{{rp|34}} They found that the lowest level of jitter to be audible was around 10&nbsp;ns ([[root mean square|rms]]). This was on a 17&nbsp;kHz [[sine wave]] test signal. With music, no listeners found jitter audible at levels lower than 20&nbsp;ns. A paper by Ashihara et al. (2005) attempted to determine the detection thresholds for random jitter in music signals. Their method involved [[ABX test|ABX listening tests]]. When discussing their results, the authors commented that:

<blockquote>
So far, actual jitter in consumer products seems to be too small to be detected at least for reproduction of music signals. It 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 the ear drums with smaller distortions than loudspeakers.<ref>{{cite journal |periodical=Acoustical Science and Technology |volume=26 |date=2005 |issue=1 |pages=50–54 |doi=10.1250/ast.26.50 |title=Detection threshold for distortions due to jitter on digital audio |first1=Kaoru |last1=Ashihara |first2=Shogo |last2=Kiryu |first3=Nobuo |last3=Koizumi |first4=Akira |last4=Nishimura |first5=Juro |last5=Ohga |first6=Masaki |last6=Sawaguchi |first7=Shokichiro |last7=Yoshikawa |url=http://www.jstage.jst.go.jp/article/ast/26/1/26_50/_article |archive-url=https://web.archive.org/web/20090812061231/http://www.jstage.jst.go.jp/article/ast/26/1/26_50/_article |archive-date=2009-08-12 |access-date=2014-01-31 |doi-access=free }}</ref>
</blockquote>

==Signal processing==
After initial recording, it is common for the audio signal to be altered in some way, such as with the use of [[Dynamic range compression|compression]], [[Equalization (audio)|equalization]], [[Delay (audio effect)|delays]] and [[reverb]]. With analog, this comes in the form of [[Outboard gear|outboard hardware components]], and with digital, the same is typically accomplished with [[audio plug-in|plug-ins]] in a [[digital audio workstation]] (DAW).

A [[Digital filter#Comparison of analog and digital filters|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.

===Analog hardware===
[[File:Phase shift.svg|thumb|alt=An illustration of phase shift.|Phase shift: the [[Sine wave|sinusoidal]] wave in red has been delayed in time equal to the angle <math>\scriptstyle \theta\,</math>, shown as the sinusoidal wave in blue.]]

When altering 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.<ref>{{citation |url=https://d3uzseaevmutz1.cloudfront.net/pubs/whitePaper/DS668WP1.pdf |title=A New Perspective on Decimation and Interpolation Filters |author=Steve Green |publisher=[[Cirrus Logic]] |access-date=2022-02-20}}</ref>

===Digital filters===
Because the variables involved can be precisely specified in the calculations, [[digital filters]] can be made to objectively perform better than analog components.<ref name="dspguide21"/><ref>{{cite book|author=John Eargle, Chris Foreman|url=https://books.google.com/books?id=YWzZe6z4xdAC&pg=PA44|title=Audio Engineering for Sound Reinforcement, The Advantages of Digital Transmission and Signal Processing|year = 2002| publisher=Hal Leonard Corporation |isbn = 9780634043550|access-date=2012-09-14}}</ref> 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 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 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.

===Analog modeling===
Many plug-ins exist now that incorporate analog modeling. There are [[audio engineer]]s 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 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".<ref>{{cite web |url=http://www.soundonsound.com/sos/may07/articles/cla.htm |title=Secrets Of The Mix Engineers: Chris Lord-Alge |date=May 2007 |access-date=2012-09-13}}</ref>

==Sound quality==

=== Subjective evaluation ===
Subjective evaluation attempts to measure how well an audio component performs according to the human ear. The most common form of subjective test 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 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 [[blinded experiment|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 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 a certain amount of time for listener training.<ref name="Borwick1994">{{cite book |last=Toole |first=Floyd |date=1994 |title=The Loudspeaker and Headphone Handbook |edition=2 |editor-first=John |editor-last=Borwick |chapter=Section 11.7: Experimental Procedure |pages=481–488 |publisher=Focal Press |isbn=0-240-51371-1}}</ref>

===Early digital recordings===
Early digital audio machines had disappointing results, with digital converters introducing errors that the ear could detect.<ref name="Watkinson1994">{{cite book |last=Watkinson |first=J. |date=1994 |title=An Introduction to Digital Audio |url=https://archive.org/details/Digital_Audio_An_Introduction_to |chapter=Section 1.2: What is digital audio? What can we hear? |pages=[https://archive.org/details/Digital_Audio_An_Introduction_to/page/n10 3], 26 |publisher=Focal Press |isbn=0-240-51378-9}}</ref> 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 was noticed that CD was far more revealing of the acoustics and ambient background noise of the recording environment.<ref name="Greenfield1986">{{cite book |last=Greenfield |first=E.|display-authors=et al |date=1986 |title=[[Penguin Guide to Compact Discs|The Penguin Guide to Compact Discs, Cassettes and LPs]] |editor-first=Ivan |editor-last=March |publisher=Penguin Books, England}}</ref> For this reason, recording techniques developed for analog disc, e.g., microphone placement, needed to be adapted to suit the new digital format.<ref name="Greenfield1986"/>

Some analog recordings were remastered for digital formats. Analog recordings made in natural concert hall acoustics tended to benefit from remastering.<ref name="Greenfield1990">{{cite book |last=Greenfield |first=E.|display-authors=et al |date=1990 |title=The Penguin Guide to Compact Discs |editor-first=Ivan |editor-last=March |chapter=Preface |pages=viii-ix |publisher=Penguin Books, England |isbn=0-14-046887-0}}</ref> The remastering process was occasionally criticised for being poorly handled. When the original analog recording was fairly bright, remastering sometimes resulted in an unnatural treble emphasis.<ref name="Greenfield1990"/>

===Super Audio CD and DVD-Audio===
The [[Super Audio CD]] (SACD) format was created by [[Sony]] and [[Philips]], who were also the developers of the earlier standard audio CD format. SACD uses [[Direct Stream Digital]] (DSD) based on [[delta-sigma modulation]]. Using this technique, the audio data is stored as a sequence of fixed amplitude (i.e. 1-bit) values at a sample rate of 2.884&nbsp;MHz, which is 64 times the 44.1&nbsp;kHz sample rate used by CD. At any point in time, the amplitude of the original analog signal is represented by the density of 1's or 0's in the data stream. This digital data stream can therefore be converted to analog by passing it through an analog low-pass filter. 

The [[DVD-Audio]] format uses standard, [[linear PCM]] at variable sampling rates and bit depths, which at the very least match and usually greatly surpass those of standard [[CD audio]] (16 bits, 44.1&nbsp;kHz).

In the popular Hi-Fi press, it had been suggested that linear PCM "creates [a] stress reaction in people", and that DSD "is the only digital recording system that does not [...] have these effects".<ref name="Hawksford2001">{{cite conference|last=Hawksford |first=M. |date=2001 |archive-date=2006-05-13 |archive-url=https://web.archive.org/web/20060513223001/http://www.essex.ac.uk/ese/research/audio_lab/malcolmspubdocs/C115%20SDM%20versus%20LPCM.pdf |url=http://www.essex.ac.uk:80/ese/research/audio_lab/malcolmspubdocs/C115%20SDM%20versus%20LPCM.pdf |title=SDM versus LPCM: The Debate Continues |conference=110th AES Convention |postscript=paper 5397 |url-status=dead }}</ref> This claim appears to originate from a 1980 article by Dr [[John Diamond (doctor)|John Diamond]].<ref>{{cite web |author=Diamond |first=John |author-link=John Diamond (doctor) |year=2003 |title=Human Stress Provoked by Digitalized Recordings |url=http://www.diamondcenter.net/digitalstress.html |archive-url=https://web.archive.org/web/20040812163148/http://www.diamondcenter.net/digitalstress.html |archive-date=2004-08-12 |access-date=17 July 2013 |work=The Diamond Center |orig-year=1980}}</ref> The core of the claim that PCM recordings (the only digital recording technique available at the time) created a stress reaction rested on using the pseudoscientific technique of [[applied kinesiology]], for example by Dr Diamond at an [[Audio Engineering Society|AES]] 66th Convention (1980) presentation with the same title.<ref>{{cite journal|last1=Diamond |first1=John |last2=Lagadec |first2=Roger |journal=[[Journal of the Audio Engineering Society]] |volume=33 |issue=12 |pages=968 |date=December 1985 |url=http://www.aes.org/e-lib/browse.cfm?elib=4408 |title=More on -Human Stress Provoked by Digitalized Recordings- and Reply |publisher=[[Audio Engineering Society|AES]] |access-date=2013-08-16}}</ref> Diamond had previously used a similar technique to demonstrate that rock music (as opposed to classical) was bad for your health due to the presence of the "stopped anapestic beat".<ref>{{cite book |title=Are the Kids All Right?: The Rock Generation and Its Hidden Death Wish |first=John Grant |last=Fuller |date=1981 |isbn=0812909704 |pages=[https://archive.org/details/arekidsallright00full/page/130 130–135] |url-access=registration |url=https://archive.org/details/arekidsallright00full/page/130 }}</ref> Diamond's claims regarding digital audio were taken up by [[Mark Levinson (audio equipment designer)|Mark Levinson]], who asserted that while PCM recordings resulted in a stress reaction, DSD recordings did not.<ref>{{cite web |title=Re-vitalizing Audio Industry:Music and Health |first=Mark |last=Levinson |url=http://www.acoust.rise.waseda.ac.jp/1bitcons/1bitforum2002/Mark.pdf |archive-url=https://web.archive.org/web/20140323004604/http://www.acoust.rise.waseda.ac.jp/1bitcons/1bitforum2002/Mark.pdf |archive-date=2014-03-23}}</ref><ref>{{cite web |url=http://www.redrosemusic.com/essay.shtml |title=Mark Levinson: CD vs. SACD and LP |first=Mark |last=Levinson |publisher=Redrosemusic.com |archive-url=https://web.archive.org/web/20120229001132/http://www.redrosemusic.com/essay.shtml |archive-date=2012-02-29 |access-date=2013-08-16}}</ref><ref>{{cite web |url=http://www.stereophile.com/images/newsletter/705Astph.html |department=Stereophilia |title=Mark Levinson & the Bobcat |first=Wes |last=Phillips |publisher=Stereophile.com |date=2005-07-05 |access-date=2013-08-16}}</ref> However, a [[double-blind]] subjective test between high resolution linear PCM (DVD-Audio) and DSD did not reveal a statistically significant difference. Listeners involved in this test noted their great difficulty in hearing any difference between the two formats.<ref>{{cite conference |conference=AES Convention:116 |location=Berlin |url=http://www.aes.org/e-lib/browse.cfm?elib=12799 |title=DVD-Audio versus SACD: Perceptual Discrimination of Digital Audio Coding Formats |first1=Dominik |last1=Blech |first2=Min-Chi |last2=Yang |publisher=[[Audio Engineering Society]] |date=8–11 May 2004 |access-date=27 July 2014 |url-status=dead |archive-url=https://web.archive.org/web/20070927200955/http://www.hfm-detmold.de/eti/projekte/diplomarbeiten/dsdvspcm/aes_paper_6086.pdf |archive-date=27 September 2007 }}</ref>

===Analog preference===
The [[vinyl revival]] is in part because of analog audio's imperfection, which adds "warmth".<ref name="deffes20200130">{{Cite news |url=https://www.theadvocate.com/baton_rouge/entertainment_life/music/article_3bcdf140-1cfd-11ea-8dd4-0778f562aac6.html |title=Repeat performance: Music lovers warming up to vinyl -- again |last=Deffes |first=Olivia |date=2020-01-30 |work=The Advocate |access-date=2020-01-30 |language=en}}</ref> Some listeners prefer such audio over that of a CD. Founder and editor Harry Pearson of ''[[The Absolute Sound]]'' magazine says that "LPs are decisively more musical. CDs drain the soul from music. The emotional involvement disappears".  Dub producer [[Adrian Sherwood]] has similar feelings about the analog cassette tape, which he prefers because of its "warmer" sound.<ref>{{cite news |author=Paul |first=James |date=2003-09-26 |title=Last night a mix tape saved my life &#124; Music &#124; The Guardian |url=http://arts.guardian.co.uk/fridayreview/story/0,12102,1049363,00.html |access-date=2012-08-15 |publisher=Arts.guardian.co.uk |location=London}}</ref>

Those who favor the digital format point to the results of blind tests, which demonstrate the high performance possible with digital recorders.<ref>{{cite web |url=http://www.bostonaudiosociety.org/bas_speaker/abx_testing2.htm |title=ABX Testing article |publisher=Boston Audio Society |date=1984-02-23 |access-date=2012-08-15}}</ref> The assertion is that the "analog sound" is more a product of analog format inaccuracies than anything else. One of the first and largest supporters of digital audio was the classical conductor [[Herbert von Karajan]], who said that digital recording was "definitely superior to any other form of recording we know". He also pioneered the unsuccessful [[Digital Compact Cassette]] and conducted the first recording ever to be commercially released on CD: Richard Strauss's [[Alpine Symphony|''Eine Alpensinfonie'']]. The perception of analog audio being demonstrably superior was also called into question by music analysts following revelations that audiophile label [[Mobile Fidelity Sound Lab]] had been covertly using [[Direct Stream Digital]] files to produce vinyl releases marketed as coming from analog master tapes, with lawyer and audiophile Randy Braun stating that "These people who claim they have golden ears and can hear the difference between analog and digital, well, it turns out you couldn't."<ref>{{cite news |author1=Edgers |first=Geoff |date=2022-08-05 |title=How a Phoenix record store owner set the audiophile world on fire |url=https://www.washingtonpost.com/music/2022/08/05/mofi-records-analog-digital-scandal/ |newspaper=[[The Washington Post]] |place=Washington, D.C. |issn=0190-8286 |oclc=1330888409}}</ref><ref name="Sinclair 2022">{{cite web |last1=Sinclair |first1=Paul |title=Saturday Deluxe / 6 August 2022 |url=https://superdeluxeedition.com/feature/saturday-deluxe-6-august-2022/ |website=SuperDeluxeEdition |access-date=9 August 2022 |ref=Sinclair 2022 |archive-url=https://web.archive.org/web/20220809195424/https://superdeluxeedition.com/feature/saturday-deluxe-6-august-2022/ |archive-date=9 August 2022 |date=6 August 2022 |url-status=live}}</ref>

==Hybrid systems==
While the words ''analog audio'' usually imply that the sound is described using a continuous signal approach, and the words ''digital audio'' imply a discrete approach, there are methods of encoding audio that fall somewhere between the two. Indeed, all analog systems show discrete (quantized) behaviour at the microscopic scale.<ref>{{cite web |url=http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/iandm/part12/page1.html |title=Analog or Digital? |publisher=St-andrews.ac.uk |access-date=2012-08-15 |last=Lesurf |first=Jim |work=The Scots Guide to Electronics}}</ref> While vinyl records and common compact cassettes are analog media and use quasi-linear physical encoding methods (e.g. spiral groove depth, [[Magnetic tape|tape]] [[magnetic field]] strength) without noticeable quantization or aliasing, there are ''analog'' non-linear systems that exhibit effects similar to those encountered on digital ones, such as aliasing and "hard" dynamic floors (e.g. frequency-modulated hi-fi audio on videotapes<!--Not just VHS Hi-Fi, but also BetaHiFi and Video Hi8, use FM audio-->, [[Pulse-width modulation|PWM]] encoded signals).

==See also==
* [[Audiophile]]
* [[Audio system measurements]]
* [[History of sound recording]]

==References==
{{reflist}}

==Bibliography==
* {{cite magazine |last=Libbey |first=Ted |url=http://nest.machinecode.org/intricate/omni_magazine/Omni%20v17%20%235%20February%201995.html |title=Digital versus analog: digital music on CD reigns as the industry standard |magazine=[[Omni (magazine)|Omni]] |date=February 1995 |volume=17 |issue=5 |access-date=22 March 2017 |archive-date=7 March 2015 |archive-url=https://web.archive.org/web/20150307064638/http://nest.machinecode.org/intricate/omni_magazine/Omni%20v17%20%235%20February%201995.html |url-status=dead }}
* Pohlmann, K. (2005). ''Principles of Digital Audio'' 5th edn, McGraw-Hill Comp.

==External links==
*{{Skeptoid | id=4303 | number=303 | title=Are Vinyl Recordings Better than Digital?}}

{{Music technology}}

{{DEFAULTSORT:Comparison Of Analog And Digital Recording}}
[[Category:Digital audio recording]]
[[Category:Sound recording]]
[[Category:Technological comparisons|Analog and digital recording]]