Difference between revisions of "Sample"
Brad Johnson (talk | contribs) (Created page with "==Overview== The term "<nowiki>Sample</nowiki>" is used to describe a method where a constantly changing input is "observed" at one instant in time. It can also be used as a noun...") |
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==Overview== | ==Overview== | ||
| − | The term "<nowiki>Sample</nowiki>" is used to describe a method where a constantly changing input is "observed" at one instant in time. It can also be used as a noun to describe the result of | + | The term "<nowiki>Sample</nowiki>" is used to describe a method where a constantly changing input is "observed" at one instant in time. It can also be used as a noun to describe the result of ''sampling''. |
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==Basics== | ==Basics== | ||
| − | + | Digital audio is somewhat analogous to film or video in that it consists of a continuous series of "still images" of the ''constantly changing'' original. In the cases of film or video; the human brain processes the incoming visual information in a way that "integrates" the rapid sequence of still pictures in a manner that is perceived as motion. This is why film actually "works" with a frame rate as low as 24 frames per second. | |
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| + | Audio is very different in that the human brain/mind can discriminate changes at much higher frequencies. In the case of digital audio; it is the output filters of the DA converter that integrate the individual samples into a very close approximation of the original voltage waveform. | ||
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| + | But how does the analog to digital converter "stop" a continuously changing waveform so it can measure the voltage? By using a [[sample and hold]] circuit. The sample and hold circuit is the digital audio equivalent of a movie camera's shutter, and like the constant speed of the movie camera's shutter being 24 fps; the AD converter's sample and hold circuit must take a "snapshot" of the audio waveform voltage once every [[sample period]]. | ||
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| + | Thus; a digital audio recording is a continuous series of ''samples'' of the constantly changing analog audio input. The samples must be played back in the same order and at the same "speed" they were recorded- the [[sample frequency]]. | ||
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| − | + | [[Category:Terminology]] | |
| + | [[Category:Audio conversion]] | ||
Latest revision as of 16:53, 29 February 2012
Overview
The term "Sample" is used to describe a method where a constantly changing input is "observed" at one instant in time. It can also be used as a noun to describe the result of sampling.
Basics
Digital audio is somewhat analogous to film or video in that it consists of a continuous series of "still images" of the constantly changing original. In the cases of film or video; the human brain processes the incoming visual information in a way that "integrates" the rapid sequence of still pictures in a manner that is perceived as motion. This is why film actually "works" with a frame rate as low as 24 frames per second.
Audio is very different in that the human brain/mind can discriminate changes at much higher frequencies. In the case of digital audio; it is the output filters of the DA converter that integrate the individual samples into a very close approximation of the original voltage waveform.
But how does the analog to digital converter "stop" a continuously changing waveform so it can measure the voltage? By using a sample and hold circuit. The sample and hold circuit is the digital audio equivalent of a movie camera's shutter, and like the constant speed of the movie camera's shutter being 24 fps; the AD converter's sample and hold circuit must take a "snapshot" of the audio waveform voltage once every sample period.
Thus; a digital audio recording is a continuous series of samples of the constantly changing analog audio input. The samples must be played back in the same order and at the same "speed" they were recorded- the sample frequency.