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

webrtc/modules/audio_processing/agc2/gain_controller2.cc

Index: webrtc/modules/audio_processing/agc2/gain_controller2.cc
diff --git a/webrtc/modules/audio_processing/agc2/gain_controller2.cc b/webrtc/modules/audio_processing/agc2/gain_controller2.cc
new file mode 100644
index 0000000000000000000000000000000000000000..d0b1b39b7f2883b86a7bbdb1f7c1fc238418af79
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+++ b/webrtc/modules/audio_processing/agc2/gain_controller2.cc
@@ -0,0 +1,65 @@
+/*
+ *  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/gain_controller2.h"
+
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kGain = 0.5f;
+
+}  // namespace
+
+int GainController2::instance_count_ = 0;
+
+GainController2::GainController2(int sample_rate_hz)
+    : sample_rate_hz_(sample_rate_hz),
+      data_dumper_(new ApmDataDumper(
+        rtc::AtomicOps::Increment(&instance_count_))),
+      digital_gain_applier_(),
+      gain_(kGain) {
+  RTC_DCHECK(sample_rate_hz_ == AudioProcessing::kSampleRate8kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate16kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate32kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate48kHz);
+  data_dumper_->InitiateNewSetOfRecordings();
+  data_dumper_->DumpRaw("gain_", 1, &gain_);
+}
+
+GainController2::~GainController2() = default;
+
+void GainController2::Process(AudioBuffer* audio) {
+  for (size_t k = 0; k < audio->num_channels(); ++k) {
+    auto channel_view = rtc::ArrayView<float>(
+        audio->channels_f()[k], audio->num_frames());
+    digital_gain_applier_.Process(gain_, channel_view);
+  }
+}
+
+bool GainController2::Validate(
+    const AudioProcessing::Config::GainController2& config) {
+  return true;
+}
+
+std::string GainController2::ToString(
+    const AudioProcessing::Config::GainController2& config) {
+  std::stringstream ss;
+  ss << "{"
+     << "enabled: " << (config.enabled ? "true" : "false") << "}";
+  return ss.str();
+}
+
+}  // namespace webrtc