commit b8dc7fa5a6d86cea2d2bedb7085617f17c9c4615
author Hanna Silen <silen@webrtc.org> Thu May 20 17:37:56 2021 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu May 20 16:53:59 2021 +0000
tree 5444eb2f774c650f6c0f0bf526c268c7392330b0
parent e2b9fc6909a5d7a951f84375622d2718eba69811

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.cc

diff --git a/modules/audio_processing/include/audio_processing.cc b/modules/audio_processing/include/audio_processing.cc
index f50a283..29dcd66 100644
--- a/modules/audio_processing/include/audio_processing.cc
+++ b/modules/audio_processing/include/audio_processing.cc
@@ -77,7 +77,11 @@
          analog_lhs.startup_min_volume == analog_rhs.startup_min_volume &&
          analog_lhs.clipped_level_min == analog_rhs.clipped_level_min &&
          analog_lhs.enable_digital_adaptive ==
-             analog_rhs.enable_digital_adaptive;
+             analog_rhs.enable_digital_adaptive &&
+         analog_lhs.clipped_level_step == analog_rhs.clipped_level_step &&
+         analog_lhs.clipped_ratio_threshold ==
+             analog_rhs.clipped_ratio_threshold &&
+         analog_lhs.clipped_wait_frames == analog_rhs.clipped_wait_frames;
 }
 
 bool Agc2Config::AdaptiveDigital::operator==(
@@ -157,6 +161,12 @@
       << gain_controller1.analog_gain_controller.clipped_level_min
       << ", enable_digital_adaptive: "
       << gain_controller1.analog_gain_controller.enable_digital_adaptive
+      << ", clipped_level_step: "
+      << gain_controller1.analog_gain_controller.clipped_level_step
+      << ", clipped_ratio_threshold: "
+      << gain_controller1.analog_gain_controller.clipped_ratio_threshold
+      << ", clipped_wait_frames: "
+      << gain_controller1.analog_gain_controller.clipped_wait_frames
       << " }}, gain_controller2: { enabled: " << gain_controller2.enabled
       << ", fixed_digital: { gain_db: "
       << gain_controller2.fixed_digital.gain_db

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;