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diff --git a/webrtc/modules/audio_processing/agc2/saturation_protector.cc b/webrtc/modules/audio_processing/agc2/saturation_protector.cc
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+++ b/webrtc/modules/audio_processing/agc2/saturation_protector.cc
@@ -0,0 +1,121 @@
+/*
+ * 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/saturation_protector.h"
+
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kMinLevelDbfs = -90.f;
+
+// Min/max margins are based on speech crest-factor.
+constexpr float kMinMarginDb = 12.f;
+constexpr float kMaxMarginDb = 25.f;
+
+using saturation_protector_impl::RingBuffer;
+
+} // namespace
+
+bool RingBuffer::operator==(const RingBuffer& b) const {
+ RTC_DCHECK_LE(size_, buffer_.size());
+ RTC_DCHECK_LE(b.size_, b.buffer_.size());
+ if (size_ != b.size_) {
+ return false;
+ }
+ for (int i = 0, i0 = FrontIndex(), i1 = b.FrontIndex(); i < size_;
+ ++i, ++i0, ++i1) {
+ if (buffer_[i0 % buffer_.size()] != b.buffer_[i1 % b.buffer_.size()]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+void RingBuffer::Reset() {
+ next_ = 0;
+ size_ = 0;
+}
+
+void RingBuffer::PushBack(float v) {
+ RTC_DCHECK_GE(next_, 0);
+ RTC_DCHECK_GE(size_, 0);
+ RTC_DCHECK_LT(next_, buffer_.size());
+ RTC_DCHECK_LE(size_, buffer_.size());
+ buffer_[next_++] = v;
+ if (rtc::SafeEq(next_, buffer_.size())) {
+ next_ = 0;
+ }
+ if (rtc::SafeLt(size_, buffer_.size())) {
+ size_++;
+ }
+}
+
+absl::optional<float> RingBuffer::Front() const {
+ if (size_ == 0) {
+ return absl::nullopt;
+ }
+ RTC_DCHECK_LT(FrontIndex(), buffer_.size());
+ return buffer_[FrontIndex()];
+}
+
+bool SaturationProtectorState::operator==(
+ const SaturationProtectorState& b) const {
+ return margin_db == b.margin_db && peak_delay_buffer == b.peak_delay_buffer &&
+ max_peaks_dbfs == b.max_peaks_dbfs &&
+ time_since_push_ms == b.time_since_push_ms;
+}
+
+void ResetSaturationProtectorState(float initial_margin_db,
+ SaturationProtectorState& state) {
+ state.margin_db = initial_margin_db;
+ state.peak_delay_buffer.Reset();
+ state.max_peaks_dbfs = kMinLevelDbfs;
+ state.time_since_push_ms = 0;
+}
+
+void UpdateSaturationProtectorState(float speech_peak_dbfs,
+ float speech_level_dbfs,
+ SaturationProtectorState& state) {
+ // Get the max peak over `kPeakEnveloperSuperFrameLengthMs` ms.
+ state.max_peaks_dbfs = std::max(state.max_peaks_dbfs, speech_peak_dbfs);
+ state.time_since_push_ms += kFrameDurationMs;
+ if (rtc::SafeGt(state.time_since_push_ms, kPeakEnveloperSuperFrameLengthMs)) {
+ // Push `max_peaks_dbfs` back into the ring buffer.
+ state.peak_delay_buffer.PushBack(state.max_peaks_dbfs);
+ // Reset.
+ state.max_peaks_dbfs = kMinLevelDbfs;
+ state.time_since_push_ms = 0;
+ }
+
+ // Update margin by comparing the estimated speech level and the delayed max
+ // speech peak power.
+ // TODO(alessiob): Check with aleloi@ why we use a delay and how to tune it.
+ const float delayed_peak_dbfs =
+ state.peak_delay_buffer.Front().value_or(state.max_peaks_dbfs);
+ const float difference_db = delayed_peak_dbfs - speech_level_dbfs;
+ if (difference_db > state.margin_db) {
+ // Attack.
+ state.margin_db =
+ state.margin_db * kSaturationProtectorAttackConstant +
+ difference_db * (1.f - kSaturationProtectorAttackConstant);
+ } else {
+ // Decay.
+ state.margin_db = state.margin_db * kSaturationProtectorDecayConstant +
+ difference_db * (1.f - kSaturationProtectorDecayConstant);
+ }
+
+ state.margin_db =
+ rtc::SafeClamp<float>(state.margin_db, kMinMarginDb, kMaxMarginDb);
+}
+
+} // namespace webrtc