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

webrtc/modules/audio_processing/agc2/gain_controller2_unittest.cc

Index: webrtc/modules/audio_processing/agc2/gain_controller2_unittest.cc
diff --git a/webrtc/modules/audio_processing/agc2/gain_controller2_unittest.cc b/webrtc/modules/audio_processing/agc2/gain_controller2_unittest.cc
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index 0000000000000000000000000000000000000000..ea945983cd5fd6e7fb34e64edd3a10587809215d
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+++ b/webrtc/modules/audio_processing/agc2/gain_controller2_unittest.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 <memory>
+#include <string>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/agc2/gain_controller2.h"
+#include "webrtc/modules/audio_processing/agc2/digital_gain_applier.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+constexpr size_t kNumFrames = 480u;
+constexpr size_t kStereo = 2u;
+
+void SetAudioBufferSamples(float value, AudioBuffer* ab) {
+  for (size_t k = 0; k < ab->num_channels(); ++k) {
+    auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames());
+    for (auto& sample : channel) { sample = value; }
+  }
+}
+
+template<typename Functor>
+bool CheckAudioBufferSamples(Functor validator, AudioBuffer* ab) {
+  for (size_t k = 0; k < ab->num_channels(); ++k) {
+    auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames());
+    for (auto& sample : channel) { if (!validator(sample)) { return false; } }
+  }
+  return true;
+}
+
+bool TestDigitalGainApplier(float sample_value, float gain, float expected) {
+  AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
+  SetAudioBufferSamples(sample_value, &ab);
+
+  DigitalGainApplier gain_applier;
+  for (size_t k = 0; k < ab.num_channels(); ++k) {
+    auto channel_view = rtc::ArrayView<float>(
+        ab.channels_f()[k], ab.num_frames());
+    gain_applier.Process(gain, channel_view);
+  }
+
+  auto check_expectation = [expected](float sample) {
+      return sample == expected; };
+  return CheckAudioBufferSamples(check_expectation, &ab);
+}
+
+}  // namespace
+
+TEST(GainController2, Instance) {
+  std::unique_ptr<GainController2> gain_controller2;
+  gain_controller2.reset(new GainController2(
+      AudioProcessing::kSampleRate48kHz));
+}
+
+TEST(GainController2, ToString) {
+  AudioProcessing::Config config;
+
+  config.gain_controller2.enabled = false;
+  EXPECT_EQ("{enabled: false}",
+            GainController2::ToString(config.gain_controller2));
+
+  config.gain_controller2.enabled = true;
+  EXPECT_EQ("{enabled: true}",
+            GainController2::ToString(config.gain_controller2));
+}
+
+TEST(GainController2, DigitalGainApplierProcess) {
+  EXPECT_TRUE(TestDigitalGainApplier(1000.0f, 0.5, 500.0f));
+}
+
+TEST(GainController2, DigitalGainApplierCheckClipping) {
+  EXPECT_TRUE(TestDigitalGainApplier(30000.0f, 1.5, 32767.0f));
+  EXPECT_TRUE(TestDigitalGainApplier(-30000.0f, 1.5, -32767.0f));
+}
+
+TEST(GainController2, Usage) {
+  std::unique_ptr<GainController2> gain_controller2;
+  gain_controller2.reset(new GainController2(
+      AudioProcessing::kSampleRate48kHz));
+  AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
+  SetAudioBufferSamples(1000.0f, &ab);
+  gain_controller2->Process(&ab);
+}
+
+}  // namespace test
+}  // namespace webrtc