Content deleted Content added
ClueBot NG (talk | contribs) m Reverting possible vandalism by 38.186.102.231 to version by Cutlass. Report False Positive? Thanks, ClueBot NG. (4392463) (Bot) |
Rescuing 4 sources and tagging 0 as dead.) #IABot (v2.0.9.5 |
||
| (2 intermediate revisions by 2 users not shown) | |||
Line 16:
Functions of space, time, or any other dimension can be sampled, and similarly in two or more dimensions.
For functions that vary with time, let <math>s(t)</math> be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring the value of the continuous function every <math>T</math> seconds, which is called the '''sampling interval''' or '''sampling period'''.<ref>{{cite book |
: <math>s(nT)</math>, for integer values of <math>n</math>.
{{anchor|Sampling rate}}The '''sampling frequency''' or '''sampling rate''', <math>f_s</math>, is the average number of samples obtained in one second, thus <math>f_s=1/T</math>, with the unit ''samples per second'', sometimes referred to as [[hertz]], for example 48 kHz is 48,000 ''samples per second''.
Line 60:
}}</ref> such as when recording music or many types of acoustic events, audio waveforms are typically sampled at 44.1 kHz ([[Compact Disc Digital Audio|CD]]), 48 kHz, 88.2 kHz, or 96 kHz.<ref>{{cite book |url=https://books.google.com/books?id=WzYm1hGnCn4C&pg=PT200 |pages=200, 446 |last=Self |first=Douglas |title=Audio Engineering Explained |publisher=Taylor & Francis US |year=2012 |isbn=978-0240812731}}</ref> The approximately double-rate requirement is a consequence of the [[Nyquist theorem]]. Sampling rates higher than about 50 kHz to 60 kHz cannot supply more usable information for human listeners. Early [[professional audio]] equipment manufacturers chose sampling rates in the region of 40 to 50 kHz for this reason.
There has been an industry trend towards sampling rates well beyond the basic requirements: such as 96 kHz and even 192 kHz<ref>{{cite web |url=http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm |title=Digital Pro Sound |access-date=8 January 2014 |archive-date=20 October 2008 |archive-url=https://web.archive.org/web/20081020231427/http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm |url-status=dead }}</ref> Even though [[Ultrasound|ultrasonic]] frequencies are inaudible to humans, recording and mixing at higher sampling rates is effective in eliminating the distortion that can be caused by [[Aliasing#Folding|foldback aliasing]]. Conversely, ultrasonic sounds may interact with and modulate the audible part of the frequency spectrum ([[intermodulation distortion]]), ''degrading'' the fidelity.<ref>{{cite journal|last=Colletti|first=Justin|date=February 4, 2013|title=The Science of Sample Rates (When Higher Is Better—And When It Isn't)|url=https://sonicscoop.com/2016/02/19/the-science-of-sample-rates-when-higher-is-better-and-when-it-isnt/?singlepage=1|journal=Trust Me I'm a Scientist|access-date=February 6, 2013|quote=in many cases, we can hear the sound of higher sample rates not because they are more transparent, but because they are less so. They can actually introduce unintended distortion in the audible spectrum}}</ref> One advantage of higher sampling rates is that they can relax the low-pass filter design requirements for [[analog-to-digital converter|ADCs]] and [[digital-to-analog converter|DACs]], but with modern oversampling [[Delta-sigma modulation|delta-sigma-converters]] this advantage is less important.
The [[Audio Engineering Society]] recommends 48 kHz sampling rate for most applications but gives recognition to 44.1 kHz for CD and other consumer uses, 32 kHz for transmission-related applications, and 96 kHz for higher bandwidth or relaxed [[anti-aliasing filter]]ing.<ref name=AES5>{{citation |url=http://www.aes.org/publications/standards/search.cfm?docID=14 |title=AES5-2008: AES recommended practice for professional digital audio – Preferred sampling frequencies for applications employing pulse-code modulation |publisher=Audio Engineering Society |year=2008 |access-date=2010-01-18}}</ref> Both Lavry Engineering and J. Robert Stuart state that the ideal sampling rate would be about 60 kHz, but since this is not a standard frequency, recommend 88.2 or 96 kHz for recording purposes.<ref>{{Cite web|url=http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf|title=The Optimal Sample Rate for Quality Audio|last=Lavry|first=Dan|date=May 3, 2012|website=Lavry Engineering Inc.|quote=Although 60 KHz would be closer to the ideal; given the existing standards, 88.2 KHz and 96 KHz are closest to the optimal sample rate.}}</ref><ref>{{Cite web|url=https://www.gearslutz.com/board/showpost.php?p=7883017&postcount=15&s=b05e50b41d1789054724882582d8351b|title=The Optimal Sample Rate for Quality Audio|last=Lavry|first=Dan|website=Gearslutz|language=en|access-date=2018-11-10|quote=I am trying to accommodate all ears, and there are reports of few people that can actually hear slightly above 20KHz. I do think that 48 KHz is pretty good compromise, but 88.2 or 96 KHz yields some additional margin.}}</ref><ref>{{Cite web|url=https://www.gearslutz.com/board/showpost.php?p=1234224&postcount=74|title=To mix at 96k or not?|last=Lavry|first=Dan|website=Gearslutz|language=en|access-date=2018-11-10|quote=Nowdays [sic] there are a number of good designers and ear people that find 60-70KHz sample rate to be the optimal rate for the ear. It is fast enough to include what we can hear, yet slow enough to do it pretty accurately.}}</ref><ref>{{Cite book|title=Coding High Quality Digital Audio|last=Stuart|first=J. Robert|date=1998|quote=both psychoacoustic analysis and experience tell us that the minimum rectangular channel necessary to ensure transparency uses linear PCM with 18.2-bit samples at 58 kHz. ... there are strong arguments for maintaining integer relationships with existing sampling rates – which suggests that 88.2 kHz or 96 kHz should be adopted.|citeseerx = 10.1.1.501.6731}}</ref>
Line 72:
|-
|5,512.5 Hz
|Supported in [[Adobe Flash|Flash]].<ref>{{Cite web |date=2013 |title=SWF File Format Specification - Version 19 |url=https://open-flash.github.io/mirrors/swf-spec-19.pdf}}</ref>
|-
| 8,000 Hz
Line 111:
|-
| 64,000 Hz
| Uncommonly used, but supported by some hardware<ref>{{Cite web|url=http://www.rme-audio.de/en/products/hdsp_9632.php|title=RME: Hammerfall DSP 9632|website=www.rme-audio.de|access-date=2018-12-18|quote=Supported sample frequencies: Internally 32, 44.1, 48, 64, 88.2, 96, 176.4, 192 kHz.}}</ref><ref>{{Cite web|url=https://www.pioneer-audiovisual.eu/uk/products/sx-s30dab|title=SX-S30DAB {{!}} Pioneer|website=www.pioneer-audiovisual.eu|access-date=2018-12-18|quote=Supported sampling rates: 44.1 kHz, 48 kHz, 64 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192 kHz|archive-date=2018-12-18|archive-url=https://web.archive.org/web/20181218145630/https://www.pioneer-audiovisual.eu/uk/products/sx-s30dab|url-status=dead}}</ref> and software.<ref>{{Cite web|url=https://steinberg.help/wavelab_pro/v9.5/en/wavelab/topics/master_section/master_section_customize_sample_rate_menu_dialog_r.html|title=Customize Sample Rate Menu|last1=Cristina Bachmann|first1=Heiko Bischoff|last2=Schütte|first2=Benjamin|website=Steinberg WaveLab Pro|language=en-US|access-date=2018-12-18|quote=Common Sample Rates: 64 000 Hz}}</ref><ref>{{Cite web|url=https://getsatisfaction.com/m-audio/topics/m-track-2x2m-cubase-pro-9-can-t-change-sample-rate|title=M Track 2x2M Cubase Pro 9 can ́t change Sample Rate|website=M-Audio|language=en-US|access-date=2018-12-18|quote=[Screenshot of Cubase]|archive-date=2018-12-18|archive-url=https://web.archive.org/web/20181218102147/https://getsatisfaction.com/m-audio/topics/m-track-2x2m-cubase-pro-9-can-t-change-sample-rate|url-status=dead}}</ref>
|-
| 88,200 Hz
Line 150:
{{See also|Audio bit depth}}
Audio is typically recorded at 8-, 16-, and 24-bit depth; which yield a theoretical maximum [[signal-to-quantization-noise ratio]] (SQNR) for a pure [[sine wave]] of, approximately; 49.93 [[Decibel|dB]], 98.09 dB, and 122.17 dB.<ref>{{cite web |url=http://www.analog.com/static/imported-files/tutorials/MT-001.pdf |title=MT-001: Taking the Mystery out of the Infamous Formula, "SNR=6.02N + 1.76dB," and Why You Should Care |access-date=2010-01-19 |archive-date=2022-10-09 |archive-url=https://ghostarchive.org/archive/20221009/http://www.analog.com/static/imported-files/tutorials/MT-001.pdf |url-status=dead }}</ref> CD quality audio uses 16-bit samples. [[Thermal noise]] limits the true number of bits that can be used in quantization. Few analog systems have [[Signal-to-noise ratio|signal to noise ratios]] (SNR) exceeding 120 dB. However, [[digital signal processing]] operations can have very high dynamic range, consequently it is common to perform mixing and mastering operations at 32-bit precision and then convert to 16- or 24-bit for distribution.
==== Speech sampling ====
Speech signals, i.e., signals intended to carry only human [[Speech communication|speech]], can usually be sampled at a much lower rate. For most [[phoneme]]s, almost all of the energy is contained in the 100 Hz – 4 kHz range, allowing a sampling rate of 8 kHz. This is the sampling rate used by nearly all [[telephony]] systems, which use the [[G.711]] sampling and quantization specifications.{{Citation needed|reason=References are needed for frequency range of human voice, and use of G.711|date=May 2018}}
=== Video sampling ===
| |||