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  2. Sampling (signal processing) - Wikipedia

    en.wikipedia.org/wiki/Sampling_(signal_processing)

    Signal sampling representation. The continuous signal S(t) is represented with a green colored line while the discrete samples are indicated by the blue vertical lines. In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples".

  3. Downsampling (signal processing) - Wikipedia

    en.wikipedia.org/wiki/Downsampling_(signal...

    Reduce high-frequency signal components with a digital lowpass filter. Decimate the filtered signal by M; that is, keep only every M th sample. Step 2 alone creates undesirable aliasing (i.e. high-frequency signal components will copy into the lower frequency band and be mistaken for lower frequencies). Step 1, when necessary, suppresses ...

  4. Delta modulation - Wikipedia

    en.wikipedia.org/wiki/Delta_modulation

    Only the change of information is sent, that is, only an increase or decrease of the signal amplitude from the previous sample is sent whereas a no-change condition causes the modulated signal to remain at the same ↗ or ↘ state of the previous sample. To achieve high signal-to-noise ratio, delta modulation must use oversampling techniques ...

  5. Differential pulse-code modulation - Wikipedia

    en.wikipedia.org/wiki/Differential_pulse-code...

    The input can be an analog signal or a digital signal. If the input is a continuous-time analog signal, it needs to be sampled first so that a discrete-time signal is the input to the DPCM encoder. Option 1: take the values of two consecutive samples; if they are analog samples, quantize them; calculate the difference between the first one and ...

  6. Nyquist–Shannon sampling theorem - Wikipedia

    en.wikipedia.org/wiki/Nyquist–Shannon_sampling...

    The Nyquist–Shannon sampling theorem is an essential principle for digital signal processing linking the frequency range of a signal and the sample rate required to avoid a type of distortion called aliasing. The theorem states that the sample rate must be at least twice the bandwidth of the signal to avoid aliasing.

  7. Pulse-code modulation - Wikipedia

    en.wikipedia.org/wiki/Pulse-code_modulation

    For each sample, one of the available values (on the y-axis) is chosen. The PCM process is commonly implemented on a single integrated circuit called an analog-to-digital converter (ADC). This produces a fully discrete representation of the input signal (blue points) that can be easily encoded as digital data for storage or manipulation.

  8. Signal processing - Wikipedia

    en.wikipedia.org/wiki/Signal_processing

    Differential equations [16] - for modeling system behavior, connecting input and output relations in linear time-invariant systems. For instance, a low-pass filter such as an RC circuit can be modeled as a differential equation in signal processing, which allows one to compute the continuous output signal as function of the input or initial ...

  9. Upsampling - Wikipedia

    en.wikipedia.org/wiki/Upsampling

    For example, if compact disc audio at 44,100 samples/second is upsampled by a factor of 5/4, the resulting sample-rate is 55,125. Fig 1: Depiction of one dot product, resulting in one output sample (in green), for the case L=4, n=9, j=3. Three conceptual "inserted zeros" are depicted between each pair of input samples.