Lesson 219

Digital Signal Processing

Sampling · Fourier · Filters

1:00

How computers turn continuous waves into numbers, analyze them in the frequency domain with the Fourier transform, and manipulate them with filters.

By the end, you can

  • Explain what sampling is and what the sample rate f<sub>s</sub> controls.
  • Apply the Nyquist–Shannon theorem to compute the minimum sampling rate for a given signal.
  • Predict the alias frequency when a tone is under-sampled, and explain why aliasing is irreversible.
  • Describe what the Fourier transform reveals and distinguish the time domain from the frequency domain.
  • Compare the FFT and the naive DFT in terms of time complexity (O(N log N) vs. O(N²)).
  • Distinguish low-pass, high-pass, and band-pass filters by which frequencies each keeps.
  • Connect filtering to convolution, and explain why CNN kernels use the same operation.
  • Estimate audio dynamic range from bit depth using the 6 dB/bit rule.
  • Identify how MP3, JPEG, and noise reduction each exploit frequency-domain processing.
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