I'm interested in how the dynamic range works in digital audio processors. What are the main stages of signal processing involved: level measurement, threshold setting, attack and release ratios, and types of compression (Hard Knee, Soft Knee)? How does the algorithm respond to fast peak signals, and what factors affect the transparency of the result? Share your experience and links to theory.
How does the dynamic range algorithm work in audio processors?
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The most vivid example of me really diving into dynamic range algorithms was when I was mixing my own demo track in my home studio. At first, I checked the RMS level meter and saw that the main material was hovering around –23 dBFS, but short peaks would jump up to –6 dBFS during transitions into the chorus parts. To avoid over-compressing, I set the threshold at –18 dB, attack at 5 ms, and release at 120 ms. By choosing a "soft knee," I got a smoother transition into compression, and even the fastest peaks would "glide" without harsh resonances, preserving the natural dynamics of the vocals.
In the same project, I experimented with a "hard knee" on the bass tracks, where more aggressive level control was needed. It turned out that with too short an attack (around 1 ms), the bass became "harsh," and artifacts were audible as "pumping noise." By increasing the attack to 10 ms and tweaking the release slightly, clarity was restored, and the compressor only caught the necessary transitions. The main takeaway for me was that clarity depends on the ratio of attack/release to the signal's character and the chosen knee type—soft knee adapts faster to quick peaks, while hard knee requires more careful timing adjustments.