AGC2 is the newer automatic gain control in libWebRTC, the adaptive digital design that replaced the original analog-loop AGC1. Its job is to keep the captured voice at a consistent target level regardless of how loud the speaker is or how far from the microphone they sit, so listeners don't have to constantly adjust their volume. AGC2 estimates the speech level and applies gain with an adaptive digital gain controller plus a limiter to catch peaks, aiming for steadier, more predictable results across the wildly varying microphones and input gains found on real devices. It works alongside the echo canceller and noise suppressor in the WebRTC processing chain, and tuning it badly causes pumping or noise breathing.
