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

webrtc/modules/audio_processing/agc2/gain_controller2.cc

+/*
+ *  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/checks.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+int GainController2::instance_count_ = 0;
+
+GainController2::GainController2(int sample_rate_hz)
+    : data_dumper_(new ApmDataDumper(instance_count_)),
+      gain_applier_(data_dumper_.get()),
+      hard_coded_gain_(0.9f) {
+  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("agc2_hard_coded_gain", 1, &hard_coded_gain_);
+  gain_applier_.Initialize(sample_rate_hz);
+  ++instance_count_;

[peah-webrtc, 2017/05/16 13:06:57]

This increement needs to be atomic as APM can be instantiated concurrently.
What about
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_)))
?

[AleBzk, 2017/05/18 08:31:37]

Thanks, great to learn this.
Then the same change should apply to level controller.
Let me know if you want that I create another CL for that change.

[peah-webrtc, 2017/05/18 10:51:18]

Good point :-) I guess the reason for that not being a problem is that the level
controller is only active on mobile devices, where only one instance is created.

If you want to create a CL for that change that would be great!

+}
+
+GainController2::~GainController2() = default;
+
+void GainController2::Process(AudioBuffer* audio) {
+  RTC_DCHECK_LT(0, audio->num_channels());
+  int num_saturations = gain_applier_.Process(hard_coded_gain_, audio);
+  data_dumper_->DumpRaw("agc2_num_saturations", 1, &num_saturations);
+}
+
+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