diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index ffe4bf47c8..c8000feaa1 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -86,16 +86,8 @@ rtc_library("biquad_filter") {
   ]
 }
 
-rtc_library("common") {
-  sources = [
-    "agc2_common.cc",
-    "agc2_common.h",
-  ]
-  deps = [
-    "../../../rtc_base:rtc_base_approved",
-    "../../../system_wrappers:field_trial",
-  ]
-  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+rtc_source_set("common") {
+  sources = [ "agc2_common.h" ]
 }
 
 rtc_library("fixed_digital") {
@@ -201,6 +193,7 @@ rtc_library("adaptive_digital_unittests") {
     "../../../rtc_base:checks",
     "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:rtc_base_approved",
+    "../../../test:test_support",
   ]
 }
 
diff --git a/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc b/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
index b55a0f5873..6f28f86378 100644
--- a/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
+++ b/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
@@ -27,9 +27,9 @@ AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
     : AdaptiveModeLevelEstimator(
           apm_data_dumper,
           AudioProcessing::Config::GainController2::LevelEstimator::kRms,
-          /*use_saturation_protector=*/true,
-          GetInitialSaturationMarginDb(),
-          GetExtraSaturationMarginOffsetDb()) {}
+          kDefaultUseSaturationProtector,
+          kDefaultInitialSaturationMarginDb,
+          kDefaultExtraSaturationMarginDb) {}
 
 AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
     ApmDataDumper* apm_data_dumper,
@@ -39,7 +39,7 @@ AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
     : AdaptiveModeLevelEstimator(apm_data_dumper,
                                  level_estimator,
                                  use_saturation_protector,
-                                 GetInitialSaturationMarginDb(),
+                                 kDefaultInitialSaturationMarginDb,
                                  extra_saturation_margin_db) {}
 
 AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
diff --git a/modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc b/modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc
index 6ab0655094..471930a612 100644
--- a/modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc
+++ b/modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc
@@ -46,8 +46,6 @@ struct TestLevelEstimator {
   std::unique_ptr<AdaptiveModeLevelEstimator> estimator;
 };
 
-}  // namespace
-
 TEST(AutomaticGainController2AdaptiveModeLevelEstimator,
      EstimatorShouldNotCrash) {
   TestLevelEstimator level_estimator;
@@ -180,4 +178,5 @@ TEST(AutomaticGainController2AdaptiveModeLevelEstimator,
             kMaxDifferenceDb);
 }
 
+}  // namespace
 }  // namespace webrtc
diff --git a/modules/audio_processing/agc2/agc2_common.cc b/modules/audio_processing/agc2/agc2_common.cc
deleted file mode 100644
index 3f697d12ac..0000000000
--- a/modules/audio_processing/agc2/agc2_common.cc
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/audio_processing/agc2/agc2_common.h"
-
-#include <stdio.h>
-
-#include <string>
-
-#include "system_wrappers/include/field_trial.h"
-
-namespace webrtc {
-
-float GetInitialSaturationMarginDb() {
-  constexpr char kForceInitialSaturationMarginFieldTrial[] =
-      "WebRTC-Audio-Agc2ForceInitialSaturationMargin";
-
-  const bool use_forced_initial_saturation_margin =
-      webrtc::field_trial::IsEnabled(kForceInitialSaturationMarginFieldTrial);
-  if (use_forced_initial_saturation_margin) {
-    const std::string field_trial_string = webrtc::field_trial::FindFullName(
-        kForceInitialSaturationMarginFieldTrial);
-    float margin_db = -1;
-    if (sscanf(field_trial_string.c_str(), "Enabled-%f", &margin_db) == 1 &&
-        margin_db >= 12.f && margin_db <= 25.f) {
-      return margin_db;
-    }
-  }
-  constexpr float kDefaultInitialSaturationMarginDb = 20.f;
-  return kDefaultInitialSaturationMarginDb;
-}
-
-float GetExtraSaturationMarginOffsetDb() {
-  constexpr char kForceExtraSaturationMarginFieldTrial[] =
-      "WebRTC-Audio-Agc2ForceExtraSaturationMargin";
-
-  const bool use_forced_extra_saturation_margin =
-      webrtc::field_trial::IsEnabled(kForceExtraSaturationMarginFieldTrial);
-  if (use_forced_extra_saturation_margin) {
-    const std::string field_trial_string = webrtc::field_trial::FindFullName(
-        kForceExtraSaturationMarginFieldTrial);
-    float margin_db = -1;
-    if (sscanf(field_trial_string.c_str(), "Enabled-%f", &margin_db) == 1 &&
-        margin_db >= 0.f && margin_db <= 10.f) {
-      return margin_db;
-    }
-  }
-  constexpr float kDefaultExtraSaturationMarginDb = 2.f;
-  return kDefaultExtraSaturationMarginDb;
-}
-}  // namespace webrtc
diff --git a/modules/audio_processing/agc2/agc2_common.h b/modules/audio_processing/agc2/agc2_common.h
index 0549898e26..c238b30881 100644
--- a/modules/audio_processing/agc2/agc2_common.h
+++ b/modules/audio_processing/agc2/agc2_common.h
@@ -53,8 +53,9 @@ constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
 constexpr float kDefaultSmoothedVadProbabilityAttack = 1.f;
 
 // Saturation Protector settings.
-float GetInitialSaturationMarginDb();
-float GetExtraSaturationMarginOffsetDb();
+constexpr bool kDefaultUseSaturationProtector = true;
+constexpr float kDefaultInitialSaturationMarginDb = 20.f;
+constexpr float kDefaultExtraSaturationMarginDb = 2.f;
 
 constexpr size_t kPeakEnveloperSuperFrameLengthMs = 400;
 static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,
diff --git a/modules/audio_processing/agc2/saturation_protector_unittest.cc b/modules/audio_processing/agc2/saturation_protector_unittest.cc
index ae75163911..2c5ee5b036 100644
--- a/modules/audio_processing/agc2/saturation_protector_unittest.cc
+++ b/modules/audio_processing/agc2/saturation_protector_unittest.cc
@@ -15,6 +15,7 @@
 #include "modules/audio_processing/agc2/agc2_common.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/gunit.h"
+#include "test/gmock.h"
 
 namespace webrtc {
 namespace {