Lesson 219
Digital Signal Processing
Sampling · Fourier · Filters
1:00How 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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