Difference between revisions of "Sample frequency"
Brad Johnson (talk | contribs) (Created page with "==Overview== In order to digitize analog audio, most contemporary systems use a system referred to as <nowiki>"sampling"</nowiki> to repeatedly measure the voltage of the analog ...") |
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==Overview== | ==Overview== | ||
| − | In order to digitize analog audio, most contemporary systems use a system referred to as <nowiki>"sampling"</nowiki> to repeatedly measure the voltage of the analog audio waveform at a regular time | + | In order to digitize analog audio, most contemporary systems use a system referred to as <nowiki>"sampling"</nowiki> to repeatedly measure the voltage of the analog audio waveform at a regular time interval. Each voltage measurement results in a binary number of a given [[wordlength]]. The series of binary “words” are typically stored consecutively in a file for later reconstruction of the analog voltage waveform by a digital to analog converter. The sample frequency is the rate at which the samples are generated and is measured in [[Hertz]] (cycles per second). The term [[sample rate]] is used interchangeably with <nowiki>sample frequency</nowiki>. |
==Basics== | ==Basics== | ||
| + | Virtually all contemporary analog audio equipment operates on the principle of an analog voltage waveform being analogous to the the original sound's air pressure <nowiki>"waveform."</nowiki> Typically; the original sound is translated from the pressure variation to electrical variations by a [[microphone]] ( a type of [[transducer]]). The resulting voltage waveform can be transmitted on wires to an amplifier and a power amplifier; then translated back into sound pressure variations by a [[speaker]]. | ||
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| + | One important consideration is how this analog waveform can be stored for later reproduction or transmission. All analog storage and transmission schemes are prone to loss of [[signal quality]]; with storage being particularly problematic. As the technology became available, [[digital audio]] systems were developed to address these issues. | ||
Revision as of 12:14, 10 January 2012
Overview
In order to digitize analog audio, most contemporary systems use a system referred to as "sampling" to repeatedly measure the voltage of the analog audio waveform at a regular time interval. Each voltage measurement results in a binary number of a given wordlength. The series of binary “words” are typically stored consecutively in a file for later reconstruction of the analog voltage waveform by a digital to analog converter. The sample frequency is the rate at which the samples are generated and is measured in Hertz (cycles per second). The term sample rate is used interchangeably with sample frequency.
Basics
Virtually all contemporary analog audio equipment operates on the principle of an analog voltage waveform being analogous to the the original sound's air pressure "waveform." Typically; the original sound is translated from the pressure variation to electrical variations by a microphone ( a type of transducer). The resulting voltage waveform can be transmitted on wires to an amplifier and a power amplifier; then translated back into sound pressure variations by a speaker.
One important consideration is how this analog waveform can be stored for later reproduction or transmission. All analog storage and transmission schemes are prone to loss of signal quality; with storage being particularly problematic. As the technology became available, digital audio systems were developed to address these issues.