Difference between revisions of "PCM"
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==Basics== | ==Basics== | ||
There are different types of encoding that can be used for digital audio, depending on the application and quality requirements. For example; professional audio requires the highest quality attainable as versus telephone communication which can employ much lower quality audio and still be acceptable for speech. One other consideration is the ease with which the digital audio signal can be processed; which in pro audio applications is important for processing such as level adjustment, [[equalization]] (or “tone” adjustment), and mixing. | There are different types of encoding that can be used for digital audio, depending on the application and quality requirements. For example; professional audio requires the highest quality attainable as versus telephone communication which can employ much lower quality audio and still be acceptable for speech. One other consideration is the ease with which the digital audio signal can be processed; which in pro audio applications is important for processing such as level adjustment, [[equalization]] (or “tone” adjustment), and mixing. | ||
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These considerations were part of the reason why PCM encoding was adopted by SONY and Phillips for the format of the encoding in the original Compact Disc (CD) standard. Computer file formats such as WAVE and AIF also utilize the PCM format for similar reasons; and most computer audio software is designed to use one of these formats as its “working” file format. | These considerations were part of the reason why PCM encoding was adopted by SONY and Phillips for the format of the encoding in the original Compact Disc (CD) standard. Computer file formats such as WAVE and AIF also utilize the PCM format for similar reasons; and most computer audio software is designed to use one of these formats as its “working” file format. | ||
The fundamental idea is similar to a “graph” of the audio waveform; where there are evenly-spaced divisions on the horizontal “X” axis that represent time, and evenly spaced divisions on the vertical “Y” axis that represent [[amplitude]]. The amplitude typically corresponds to the [[voltage]] of the input analog waveform because voltage waveform is how the analog audio waveform is represented and transmitted between audio devices (in the vast majority of examples). Each time division represents one “cycle” at the [[sample frequency]]; which corresponds to the time at which the analog signal is either [[sampled]] by the AD converter during recording or the time at which the sampled voltage is reproduced by the DA converter during playback. | The fundamental idea is similar to a “graph” of the audio waveform; where there are evenly-spaced divisions on the horizontal “X” axis that represent time, and evenly spaced divisions on the vertical “Y” axis that represent [[amplitude]]. The amplitude typically corresponds to the [[voltage]] of the input analog waveform because voltage waveform is how the analog audio waveform is represented and transmitted between audio devices (in the vast majority of examples). Each time division represents one “cycle” at the [[sample frequency]]; which corresponds to the time at which the analog signal is either [[sampled]] by the AD converter during recording or the time at which the sampled voltage is reproduced by the DA converter during playback. | ||
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Here is a simplified example: | Here is a simplified example: | ||
(Insert graphic) | (Insert graphic) | ||
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Similar to the cases of film or video recording and playback; the accuracy of ''both'' the divisions in the horizontal and vertical scales (time domain and amplitude domain) is directly related to the accuracy of the reproduction (reconstruction) of the original analog audio signal. Due to the both the differences between visual and auditory perception as well as the large difference in the frequency at which the information is <nowiki>“sampled;”</nowiki> the results of variations in the time domain | Similar to the cases of film or video recording and playback; the accuracy of ''both'' the divisions in the horizontal and vertical scales (time domain and amplitude domain) is directly related to the accuracy of the reproduction (reconstruction) of the original analog audio signal. Due to the both the differences between visual and auditory perception as well as the large difference in the frequency at which the information is <nowiki>“sampled;”</nowiki> the results of variations in the time domain | ||
Revision as of 12:39, 5 January 2012
Overview
The term "PCM" stands for "Pulse Code Modulation" and refers one system for encoding digital audio after AD conversion.
History
Basics
There are different types of encoding that can be used for digital audio, depending on the application and quality requirements. For example; professional audio requires the highest quality attainable as versus telephone communication which can employ much lower quality audio and still be acceptable for speech. One other consideration is the ease with which the digital audio signal can be processed; which in pro audio applications is important for processing such as level adjustment, equalization (or “tone” adjustment), and mixing.
These considerations were part of the reason why PCM encoding was adopted by SONY and Phillips for the format of the encoding in the original Compact Disc (CD) standard. Computer file formats such as WAVE and AIF also utilize the PCM format for similar reasons; and most computer audio software is designed to use one of these formats as its “working” file format. The fundamental idea is similar to a “graph” of the audio waveform; where there are evenly-spaced divisions on the horizontal “X” axis that represent time, and evenly spaced divisions on the vertical “Y” axis that represent amplitude. The amplitude typically corresponds to the voltage of the input analog waveform because voltage waveform is how the analog audio waveform is represented and transmitted between audio devices (in the vast majority of examples). Each time division represents one “cycle” at the sample frequency; which corresponds to the time at which the analog signal is either sampled by the AD converter during recording or the time at which the sampled voltage is reproduced by the DA converter during playback.
Here is a simplified example: (Insert graphic)
Similar to the cases of film or video recording and playback; the accuracy of both the divisions in the horizontal and vertical scales (time domain and amplitude domain) is directly related to the accuracy of the reproduction (reconstruction) of the original analog audio signal. Due to the both the differences between visual and auditory perception as well as the large difference in the frequency at which the information is “sampled;” the results of variations in the time domain