[tiredIsa]

For syrnike13-app issue #165: replace the application's hand-written gain with WebRTC AGC2.

• RealtimeAudioProcessing::processGain() — one 10 ms frame through a prewarmed APM with gain_controller2.adaptive_digital (C ABI in livekit-ffi/src/realtime_apm.rs).
• Separate from process() so the caller can take its gate measurement after AEC/NS and before gain. process() is unchanged.

Verification (local, Windows): cargo test -p livekit-ffi --lib realtime_apm::tests (2), ctest -R AudioProcessingModuleTest (33, incl. a test that quiet speech-like input is raised). App-side offline measurement with SAPI speech: quiet mic -27 → -22 dBFS, SNR unchanged.

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• docs/realtime-microphone-dsp.md
• include/livekit/realtime_audio_processing.h
• patches/client-sdk-rust/0001-native-v2-runtime.patch
• src/realtime_audio_processing.cpp
• src/tests/unit/test_audio_processing_module.cpp

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