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  1. #1
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    ! PCM - Cool Edit 2.1 - ( Frequency Analysis (Ctrl+Z)) ? sample rate = 44,1 kHz, , .. 22,050 kHz, sample rate = 96 kHz, , .. 48 kHz. .. 96 kHz, ( 18-20 kHz)? 96 kHz 20-20k Hz?

    , Cool Edit-a, ( , ):

    - 44,1 kHz 22,050 kHz
    22 050 / 44 100 = 0,5

    - 96 kHz 48,000 kHz
    48 000 / 96 000 = 0,5

    ? , , Cool Edit ?

  2. #2
    Registered User tedych's Avatar
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    , - , , . . , 1 , - 500 .
    DAC-a, - - . - , 22 44 . - DAC / ( ), . .

  3. #3
    loudnes's Avatar
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    96khz sample rate 20 - 20 ( ) 44.1khz 96 44.1 ( )
    hardware fan

  4. #4
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    loudnes .

    Cool Edit:

    FLAC 44,1 96 kHz Sample (96 FLAC, )
    http://img512.imageshack.us/img512/7053/44196.png

    44,1 96 kHz Sample
    http://img149.imageshack.us/img149/8240/441mic.png

  5. #5
    loudnes's Avatar
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    ,
    mialy , .
    hardware fan

  6. #6
    Reigsetder Ures Roland's Avatar
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    "" . - DAC- -"" noise shaping, low pass ( - ), "" .
    , , , .

    , 23 -90 , . noise shaping- low pass 25 . *( low pass- )



    , . 32 96 DAC-.
    , framerate-. - - ,
    Last edited by Roland; 19th May 2009 at 23:35.
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  7. #7
    Registered User tedych's Avatar
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    96 , . 96 , 1 . 96,000 , -. . - , -. (, ) , - , - - DAC.
    , .

  8. #8
    Registered User tedych's Avatar
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    , .
    / , ( 16), SNR- , 48 .
    .

  9. #9
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    , ( 96k Sample) 48 kHz, - ( ). , , , Cool Edit-a (.. , 44,1 kHz ).
    , .

    .. , !

  10. #10
    philosophus duratea icaci's Avatar
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    - - , . , sinc(t) = sin(t)/t, ( ). ( ) - sinc(t) . , - .

    - , , - :
    Quote Originally Posted by Wikipedia
    In practice, neither of the two statements of the sampling theorem described above can be completely satisfied, and neither can the reconstruction formula be precisely implemented. The reconstruction process that involves scaled and delayed sinc functions can be described as ideal. It cannot be realized in practice since it implies that each sample contributes to the reconstructed signal at almost all time points, requiring summing an infinite number of terms. Instead, some type of approximation of the sinc functions, finite in length, has to be used. The error that corresponds to the sinc-function approximation is referred to as interpolation error. Practical digital-to-analog converters produce neither scaled and delayed sinc functions nor ideal impulses (that if ideally low-pass filtered would yield the original signal), but a sequence of scaled and delayed rectangular pulses. This practical piecewise-constant output can be modeled as a zero-order hold filter driven by the sequence of scaled and delayed dirac impulses referred to in the mathematical basis section below. A shaping filter is sometimes used after the DAC with zero-order hold to make a better overall approximation.
    Last edited by icaci; 19th May 2009 at 17:43.
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