AGC2 as a new APM sub-module operating with hard-coded gain.

webrtc/modules/audio_processing/agc2/digital_gain_applier.cc

Index: webrtc/modules/audio_processing/agc2/digital_gain_applier.cc
diff --git a/webrtc/modules/audio_processing/agc2/digital_gain_applier.cc b/webrtc/modules/audio_processing/agc2/digital_gain_applier.cc
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index 0000000000000000000000000000000000000000..e245e948175121e4ae2db440ce96591d802fe29c
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+++ b/webrtc/modules/audio_processing/agc2/digital_gain_applier.cc
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+/*
+ *  Copyright (c) 2017 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 "webrtc/modules/audio_processing/agc2/digital_gain_applier.h"
+
+#include <algorithm>
+
+namespace webrtc {
+namespace {
+
+const float kMaxSampleValue = 32767.0f;
+const float kMinSampleValue = -32767.0f;
+
+}  // namespace
+
+DigitalGainApplier::DigitalGainApplier() = default;
+
+void DigitalGainApplier::Process(float gain, AudioBuffer* audio) {
+  if (gain == 1.f) { return; }
+  for (size_t k = 0; k < audio->num_channels(); ++k) {
+    auto channel_view = rtc::ArrayView<float>(
+        audio->channels_f()[k], audio->num_frames());
+    ApplyGain(gain, channel_view);
+    LimitToAllowedRange(channel_view);
+  }
+}
+
+void DigitalGainApplier::ApplyGain(float gain, rtc::ArrayView<float> x) {
+  for (auto& v : x) { v *= gain; }
+}
+
+void DigitalGainApplier::LimitToAllowedRange(rtc::ArrayView<float> x) {
+  for (auto& v : x) {
+    v = std::max(kMinSampleValue, v);
+    v = std::min(kMaxSampleValue, v);
+  }
+}
+
+}  // namespace webrtc