Scribd Logo
Upload

Welcome to Scribd!

  • 76 views

    Adc and Dac Lec 2 (1st)

    Uploaded by

    Pahala Samosir
    ADC and DAC allow digital computers to interact with analog signals by converting between digital and analog formats. During ADC, the analog signal is sampled and quantized, which introduces errors. The analog signal must be held steady during sampling to avoid changing. There are also errors from limited precision of digital representation and inability to detect low frequencies without long enough sampling. The sampling frequency must be at least twice the maximum signal frequency to avoid aliasing during reconstruction.

    Copyright:

    © All Rights Reserved
    Download as PPT, PDF, TXT or read online from Scribd

    Adc and Dac Lec 2 (1st)

    Uploaded by

    Pahala Samosir
    76 views10 pages
    ADC and DAC allow digital computers to interact with analog signals by converting between digital and analog formats. During ADC, the analog signal is sampled and quantized, which introduces errors. The analog signal must be held steady during sampling to avoid changing. There are also errors from limited precision of digital representation and inability to detect low frequencies without long enough sampling. The sampling frequency must be at least twice the maximum signal frequency to avoid aliasing during reconstruction.

    Copyright

    © © All Rights Reserved

    Available Formats

    PPT, PDF, TXT or read online from Scribd

    Did you find this document useful?

    Is this content inappropriate?

    ADC and DAC allow digital computers to interact with analog signals by converting between digital and analog formats. During ADC, the analog signal is sampled and quantized, which introduces errors. The analog signal must be held steady during sampling to avoid changing. There are also errors from limited precision of digital representation and inability to detect low frequencies without long enough sampling. The sampling frequency must be at least twice the maximum signal frequency to avoid aliasing during reconstruction.

    Copyright:

    © All Rights Reserved
    Download as PPT, PDF, TXT or read online from Scribd
    Download as ppt, pdf, or txt
    76 views10 pages

    Adc and Dac Lec 2 (1st)

    Uploaded by

    Pahala Samosir
    ADC and DAC allow digital computers to interact with analog signals by converting between digital and analog formats. During ADC, the analog signal is sampled and quantized, which introduces errors. The analog signal must be held steady during sampling to avoid changing. There are also errors from limited precision of digital representation and inability to detect low frequencies without long enough sampling. The sampling frequency must be at least twice the maximum signal frequency to avoid aliasing during reconstruction.

    Copyright:

    © All Rights Reserved
    Download as PPT, PDF, TXT or read online from Scribd
    Download as ppt, pdf, or txt
    You are on page 1of 10

    ADC and DAC

    Analog-to-Digital Conversion (ADC) and Digital-to-Analog


    Conversion (DAC) are the processes that allow digital
    computers to interact with these everyday signals.
    Digital information is different from its continuous counterpart
    in two important respects: it is
    sampled, and it is quantized
    http://www.dsptutor.freeuk.com/aliasing/AD102.html
    ADC
    Most DSP applications deal with analogue signals.
    the analogue signal has to be converted to digital form
    The analogue signal - a continuous variable defined with
    infinite precision - is converted to a discrete sequence of
    measured values which are represented digitally.
    Information is lost in converting from analogue to digital,
    due to:
    -inaccuracies in the measurement
    -uncertainty in timing
    -limits on the duration of the measuremen
    These effects are called quantisation errors
    ADC

    Sample and Hold : The continuous analogue signal has to be held


    before it can be sampled.
    Otherwise, the signal would be changing during the measurement.
    ADC
    In the process of sampling the signal, some information is lost
    Due to Following 4 reasons:
    Quantization Error
    • When the signal is converted to digital form, the precision is limited by the number of bits
    available.
    • The diagram shows an analogue signal which is then converted to a digital representation - in
    this case, with 8 bit precision.
    • The smoothly varying analogue signal can only be represented as a 'stepped' waveform due to
    the limited precision.
    • Sadly, the errors introduced by digitisation are both non linear and signal dependent.
    Sampling frequency
    • We cannot see slow changes in the signal if we don't wait long enough.
    • In fact we must sample for long enough to detect not only low frequencies in
    the signal, but also small differences between frequencies. The length of
    time for which we are prepared to sample the signal determines our ability
    to resolve adjacent frequencies - the frequency resolution.
    • We must sample for at least one complete cycle of the lowest frequency we
    want to resolve.

    We must sample fast to avoid and for a long time to achieve a good frequency resolution. But sampling fast
    for a long time means we will have a lot of samples - and lots of samples means lots of computation, for
    which we generally don't have time. So we will have to compromise between resolving frequency
    components of the signal, and being able to see high frequencies .
    Proper and improper sampling rate

    Aliasing due to low sampling rate


    The sampling theorem
    • states that for a limited bandwidth signal with
    maximum frequency fmax, the sampling frequency
    fs must be greater than twice of the maximum
    frequency fmax, i.e.,
    fs > 2· fmax
    • For the signal be uniquely reconstructed without
    aliasing.
    The 2· fmax is called the Nyquist sampling rate.
    The fmax, is called the Nyquist frequency.
    Do not confuse the above two values.
    Aliasing
    • For example, the human ear can detect sound across
    the frequency range of 20 Hz to 20 kHz. According to the
    sampling theorem, one should sample sound signals at
    least at 40 kHz in order for the reconstructed sound
    signal to be acceptable to the human ear. Components
    higher than 20 kHz cannot be detected, but they can still
    pollute the sampled signal through aliasing. Therefore,
    frequency components above 20 kHz are removed from
    the sound signal before sampling by a band-pass or low-
    pass analog filter. Practically speaking, the sampling rate
    is typically set at 44 kHz (rather than 40 kHz) in order to
    avoid signal contamination from the filter rolloff.

    You might also like