Computer Science, asked by Rojalin4327, 1 year ago

Natural sampling sampling of pulse code modulation

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Answered by prashanth1551
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Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audioin computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.
Pulse-code modulationFilename extension.L16, .WAV, .AIFF, .AU, .PCM[1]Internet media typeaudio/L16, audio/L8,[2] audio/L20, audio/L24[3][4]Type code"AIFF" for L16,[1] none[3]Magic numbervariesType of formatuncompressed audioContained byAudio CD, AES3, WAV, AIFF, AU, M2TS, VOB, and many othersExtended fromPCM
Linear pulse-code modulation (LPCM) is a specific type of PCM where the quantization levels are linearly uniform.[5] This is in contrast to PCM encodings where quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm). Though PCM is a more general term, it is often used to describe data encoded as LPCM.
A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that can be used to represent each sample.
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