Make AgcManagerDirect clipping parameters configurable

Bug: webrtc:12774
Change-Id: I99824b5aabe6f921a5db425dd1c1c1d4c606186c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219681
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34069}

modules/audio_processing/include/audio_processing.h

diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 6bc50f0..6622097 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -59,9 +59,9 @@
 //
 // Must be provided through AudioProcessingBuilder().Create(config).
 #if defined(WEBRTC_CHROMIUM_BUILD)
-static const int kAgcStartupMinVolume = 85;
+static constexpr int kAgcStartupMinVolume = 85;
 #else
-static const int kAgcStartupMinVolume = 0;
+static constexpr int kAgcStartupMinVolume = 0;
 #endif  // defined(WEBRTC_CHROMIUM_BUILD)
 static constexpr int kClippedLevelMin = 70;
 
@@ -334,6 +334,15 @@
         // clipping.
         int clipped_level_min = kClippedLevelMin;
         bool enable_digital_adaptive = true;
+        // Amount the microphone level is lowered with every clipping event.
+        // Limited to (0, 255].
+        int clipped_level_step = 15;
+        // Proportion of clipped samples required to declare a clipping event.
+        // Limited to (0.f, 1.f).
+        float clipped_ratio_threshold = 0.1f;
+        // Time in frames to wait after a clipping event before checking again.
+        // Limited to values higher than 0.
+        int clipped_wait_frames = 300;
       } analog_gain_controller;
     } gain_controller1;