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diff --git a/webrtc/modules/audio_processing/agc2/agc2_common.h b/webrtc/modules/audio_processing/agc2/agc2_common.h
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+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+constexpr float kMinFloatS16Value = -32768.f;
+constexpr float kMaxFloatS16Value = 32767.f;
+constexpr float kMaxAbsFloatS16Value = 32768.0f;
+
+constexpr size_t kFrameDurationMs = 10;
+constexpr size_t kSubFramesInFrame = 20;
+constexpr size_t kMaximalNumberOfSamplesPerChannel = 480;
+
+constexpr float kAttackFilterConstant = 0.f;
+
+// Adaptive digital gain applier settings below.
+constexpr float kHeadroomDbfs = 1.f;
+constexpr float kMaxGainDb = 30.f;
+constexpr float kInitialAdaptiveDigitalGainDb = 8.f;
+// At what limiter levels should we start decreasing the adaptive digital gain.
+constexpr float kLimiterThresholdForAgcGainDbfs = -kHeadroomDbfs;
+
+// This is the threshold for speech. Speech frames are used for updating the
+// speech level, measuring the amount of speech, and decide when to allow target
+// gain reduction.
+constexpr float kVadConfidenceThreshold = 0.9f;
+
+// The amount of 'memory' of the Level Estimator. Decides leak factors.
+constexpr size_t kFullBufferSizeMs = 1200;
+constexpr float kFullBufferLeakFactor = 1.f - 1.f / kFullBufferSizeMs;
+
+constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
+
+// Robust VAD probability and speech decisions.
+constexpr float kDefaultSmoothedVadProbabilityAttack = 1.f;
+constexpr int kDefaultLevelEstimatorAdjacentSpeechFramesThreshold = 1;
+
+// Saturation Protector settings.
+constexpr float kDefaultInitialSaturationMarginDb = 20.f;
+constexpr float kDefaultExtraSaturationMarginDb = 2.f;
+
+constexpr size_t kPeakEnveloperSuperFrameLengthMs = 400;
+static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,
+              "Full buffer size should be a multiple of super frame length for "
+              "optimal Saturation Protector performance.");
+
+constexpr size_t kPeakEnveloperBufferSize =
+    kFullBufferSizeMs / kPeakEnveloperSuperFrameLengthMs + 1;
+
+// This value is 10 ** (-1/20 * frame_size_ms / satproc_attack_ms),
+// where satproc_attack_ms is 5000.
+constexpr float kSaturationProtectorAttackConstant = 0.9988493699365052f;
+
+// This value is 10 ** (-1/20 * frame_size_ms / satproc_decay_ms),
+// where satproc_decay_ms is 1000.
+constexpr float kSaturationProtectorDecayConstant = 0.9997697679981565f;
+
+// This is computed from kDecayMs by
+// 10 ** (-1/20 * subframe_duration / kDecayMs).
+// |subframe_duration| is |kFrameDurationMs / kSubFramesInFrame|.
+// kDecayMs is defined in agc2_testing_common.h
+constexpr float kDecayFilterConstant = 0.9998848773724686f;
+
+// Number of interpolation points for each region of the limiter.
+// These values have been tuned to limit the interpolated gain curve error given
+// the limiter parameters and allowing a maximum error of +/- 32768^-1.
+constexpr size_t kInterpolatedGainCurveKneePoints = 22;
+constexpr size_t kInterpolatedGainCurveBeyondKneePoints = 10;
+constexpr size_t kInterpolatedGainCurveTotalPoints =
+    kInterpolatedGainCurveKneePoints + kInterpolatedGainCurveBeyondKneePoints;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_