Sampling (signal processing): Difference between revisions

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}}</ref> such as when recording music or many types of acoustic events, audio waveforms are typically sampled at 44.1&nbsp;kHz ([[Compact Disc Digital Audio|CD]]), 48&nbsp;kHz, 88.2&nbsp;kHz, or 96&nbsp;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&nbsp;kHz to 60 &nbsp;kHz cannot supply more usable information for human listeners. Early [[professional audio]] equipment manufacturers chose sampling rates in the region of 40 to 50&nbsp;kHz for this reason.
 
There has been an industry trend towards sampling rates well beyond the basic requirements: such as 96&nbsp;kHz and even 192&nbsp;kHz<ref>{{cite web|url=http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm |title=Digital Pro Sound |access-date=8 January 2014}}</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 &nbsp;kHz sampling rate for most applications but gives recognition to 44.1&nbsp;kHz for CD and other consumer uses, 32&nbsp;kHz for transmission-related applications, and 96&nbsp;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&nbsp;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 &nbsp;KHz would be closer to the ideal; given the existing standards, 88.2&nbsp;KHz and 96&nbsp;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 48KHz48&nbsp;KHz is pretty good compromise, but 88.2 or 96KHz96&nbsp;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 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 58kHz58&nbsp;kHz. ... there are strong arguments for maintaining integer relationships with existing sampling rates – which suggests that 88.2&nbsp;kHz or 96kHz96&nbsp;kHz should be adopted.|citeseerx = 10.1.1.501.6731}}</ref>
 
A more complete list of common audio sample rates is:
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'''Complex sampling''' (or '''I/Q sampling''') is the simultaneous sampling of two different, but related, waveforms, resulting in pairs of samples that are subsequently treated as [[complex numbers]].{{efn-ua|
Sample-pairs are also sometimes viewed as points on a [[constellation diagram]].
}}&nbsp; When one waveform<math>, \hat s(t),</math>&nbsp; is the [[Hilbert transform]] of the other waveform<math>, s(t),\,</math>&nbsp; the complex-valued function, &nbsp;<math>s_a(t) \triangleq s(t) + i\cdot \hat s(t),</math>&nbsp; is called an [[analytic signal]],&nbsp; whose Fourier transform is zero for all negative values of frequency. In that case, the [[Nyquist rate]] for a waveform with no frequencies ≥&nbsp;''B'' can be reduced to just ''B'' (complex samples/sec), instead of <math>2B</math> (real samples/sec).{{efn-ua|
When the complex sample-rate is ''B'', a frequency component at 0.6&nbsp;''B'', for instance, will have an alias at −0.4&nbsp;''B'', which is unambiguous because of the constraint that the pre-sampled signal was analytic. Also see {{slink|Aliasing|Complex sinusoids}}.
}} More apparently, the [[Baseband#Equivalent baseband signal|equivalent baseband waveform]], &nbsp;<math>s_a(t)\cdot e^{-i 2\pi \frac{B}{2} t},</math>&nbsp; also has a Nyquist rate of <math>B,</math> because all of its non-zero frequency content is shifted into the interval <math>[-B/2,B/2].</math>
 
Although complex-valued samples can be obtained as described above, they are also created by manipulating samples of a real-valued waveform. For instance, the equivalent baseband waveform can be created without explicitly computing <math>\hat s(t),</math>&nbsp; by processing the product sequence<math>, \left [s(nT)\cdot e^{-i 2 \pi \frac{B}{2}Tn}\right ],</math>{{efn-ua|
When ''s''(''t'') is sampled at the Nyquist frequency (1/''T'' {{=}} 2''B''), the product sequence simplifies to <math>\left [s(nT)\cdot (-i)^n\right ].</math>
}} &nbsp;through a digital low-pass filter whose cutoff frequency is <math>B/2.</math>{{efn-ua|