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In the field of digital signal processing, the sampling theorem is a fundamental bridge between continuous-time signals (often called "analog signals") and discrete-time signals (often called "digital signals"). It establishes a sufficient condition for a sample rate that permits a discrete sequence of ''samples'' to capture all the information from a continuous-time signal of finite bandwidth. Strictly speaking, the theorem only applies to a class of mathematical functions having a Fourier transform that is zero outside of a finite region of frequencies (see Fig 1). Intuitively we expect that when one reduces a continuous function to a discrete sequence and interpolates back to a continuous function, the fidelity of the result depends on the density (or sample rate) of the original samples. The sampling theorem introduces the concept of a sample rate that is sufficient for perfect fidelity for the class of functions that are bandlimited to a given bandwidth, such that no actual information is lost in the sampling process. It expresses the sufficient sample rate in terms of the bandwidth for the class of functions. The theorem also leads to a formula for perfectly reconstructing the original continuous-time function from the samples. Perfect reconstruction may still be possible when the sample-rate criterion is not satisfied, provided other constraints on the signal are known. (See below, and Compressed sensing.) In some cases (when the sample-rate criterion is not satisfied), utilizing additional constraints allows for approximate reconstructions. The fidelity of these reconstructions can be verified and quantified utilizing Bochner's theorem. The name ''Nyquist–Shannon sampling theorem'' honors Harry Nyquist and Claude Shannon. The theorem was also discovered independently by E. T. Whittaker, by Vladimir Kotelnikov, and by others. It is thus also known by the names ''Nyquist–Shannon–Kotelnikov'', ''Whittaker–Shannon–Kotelnikov'', ''Whittaker–Nyquist–Kotelnikov–Shannon'', and ''cardinal theorem of interpolation''. ==Introduction== Sampling is the process of converting a signal (for example, a function of continuous time and/or space) into a numeric sequence (a function of discrete time and/or space). Shannon's version of the theorem states:〔, "Communication in the presence of noise", Proc. Institute of Radio Engineers, vol. 37, no. 1, pp. 10–21, Jan. 1949. (Reprint as classic paper in: ''Proc. IEEE'', vol. 86, no. 2, (Feb. 1998) )〕 If a function x(t) contains no frequencies higher than ''B'' hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2''B'') seconds apart. A sufficient sample-rate is therefore 2''B'' samples/second, or anything larger. Equivalently, for a given sample rate ''f''s, perfect reconstruction is guaranteed possible for a bandlimit . When the bandlimit is too high (or there is no bandlimit), the reconstruction exhibits imperfections known as aliasing. Modern statements of the theorem are sometimes careful to explicitly state that ''x''(''t'') must contain no sinusoidal component at exactly frequency ''B'', or that ''B'' must be strictly less than ½ the sample rate. The two thresholds, 2''B'' and ''f''s/2 are respectively called the Nyquist rate and Nyquist frequency. And respectively, they are attributes of ''x''(''t'') and of the sampling equipment. The condition described by these inequalities is called the Nyquist criterion, or sometimes the ''Raabe condition''. The theorem is also applicable to functions of other domains, such as ''space,'' in the case of a digitized image. The only change, in the case of other domains, is the units of measure applied to ''t'', ''f''s, and ''B''. The is customarily used to represent the interval between samples and is called the sample period or sampling interval. And the samples of function ''x''(''t'') are commonly denoted by (alternatively "''xn''" in older signal processing literature), for all integer values of ''n''. The mathematically ideal way to interpolate the sequence involves the use of sinc functions, like those shown in Fig 2. Each sample in the sequence is replaced by a sinc function, centered on the time axis at the original location of the sample, ''nT'', with the amplitude of the sinc function scaled to the sample value, ''x''(). Subsequently, the sinc functions are summed into a continuous function. A mathematically equivalent method is to convolve one sinc function with a series of Dirac delta pulses, weighted by the sample values. Neither method is numerically practical. Instead, some type of approximation of the sinc functions, finite in length, is used. The imperfections attributable to the approximation are known as ''interpolation error''. Practical digital-to-analog converters produce neither scaled and delayed sinc functions, nor ideal Dirac pulses. Instead they produce a piecewise-constant sequence of scaled and delayed rectangular pulses (the zero-order hold), usually followed by an "anti-imaging filter" to clean up spurious high-frequency content. 抄文引用元・出典: フリー百科事典『 ウィキペディア（Wikipedia）』 ■ウィキペディアで「Nyquist–Shannon sampling theorem」の詳細全文を読む スポンサード リンク
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