New module for the adaptive level controlling functionality in the audio processing module

webrtc/modules/audio_processing/level_controller/level_controller.h

Index: webrtc/modules/audio_processing/level_controller/level_controller.h
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.h b/webrtc/modules/audio_processing/level_controller/level_controller.h
new file mode 100644
index 0000000000000000000000000000000000000000..cbe423a783f1c760fe1a6771329acddb17a4405f
--- /dev/null
+++ b/webrtc/modules/audio_processing/level_controller/level_controller.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/level_controller/gain_applier.h"
+#include "webrtc/modules/audio_processing/level_controller/gain_selector.h"
+#include "webrtc/modules/audio_processing/level_controller/noise_level_estimator.h"
+#include "webrtc/modules/audio_processing/level_controller/peak_level_estimator.h"
+#include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
+#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class LevelController {
+ public:
+  LevelController();
+  ~LevelController();
+
+  void Initialize(int sample_rate_hz);
+  void Process(AudioBuffer* audio);
+  float GetLastGain() { return last_gain_; }
+
+ private:
+  class Metrics {
+   public:
+    Metrics() { Initialize(AudioProcessing::kSampleRate48kHz); }
+    void Initialize(int sample_rate_hz);
+    void Update(float peak_level, float noise_level, float gain);
+
+   private:
+    void ResetEstimation();
+
+    int metrics_frame_counter_;

[hlundin-webrtc, 2016/06/29 08:56:28]

A counter should probably be size_t.

[peah-webrtc, 2016/06/29 09:13:53]

Done.

+    float gain_sum_;
+    int32_t peak_level_sum_;
+    float noise_energy_sum_;
+    float max_gain_;
+    float max_peak_level_;
+    float max_noise_energy_;
+    float frame_length_;
+  };
+
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  GainSelector gain_selector_;
+  GainApplier gain_applier_;
+  SignalClassifier signal_classifier_;
+  NoiseLevelEstimator noise_level_estimator_;
+  PeakLevelEstimator peak_level_estimator_;
+  SaturatingGainEstimator saturating_gain_estimator_;
+  Metrics metrics_;
+  rtc::Optional<int> sample_rate_hz_;
+  static int instance_count_;
+  float dc_level_[2];
+  float dc_forgetting_factor_;
+  float last_gain_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(LevelController);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_