1 files changed, 77 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_controller2.h b/third_party/libwebrtc/modules/audio_processing/gain_controller2.h
new file mode 100644
index 0000000000..304fa40489
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_controller2.h
@@ -0,0 +1,77 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+
+#include <atomic>
+#include <memory>
+#include <string>
+
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/agc2/limiter.h"
+#include "modules/audio_processing/agc2/vad_wrapper.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+// Gain Controller 2 aims to automatically adjust levels by acting on the
+// microphone gain and/or applying digital gain.
+class GainController2 {
+ public:
+  // Ctor. If `use_internal_vad` is true, an internal voice activity
+  // detector is used for digital adaptive gain.
+  GainController2(const AudioProcessing::Config::GainController2& config,
+                  int sample_rate_hz,
+                  int num_channels,
+                  bool use_internal_vad);
+  GainController2(const GainController2&) = delete;
+  GainController2& operator=(const GainController2&) = delete;
+  ~GainController2();
+
+  // Detects and handles changes of sample rate and/or number of channels.
+  void Initialize(int sample_rate_hz, int num_channels);
+
+  // Sets the fixed digital gain.
+  void SetFixedGainDb(float gain_db);
+
+  // Applies fixed and adaptive digital gains to `audio` and runs a limiter.
+  // If the internal VAD is used, `speech_probability` is ignored. Otherwise
+  // `speech_probability` is used for digital adaptive gain if it's available
+  // (limited to values [0.0, 1.0]).
+  void Process(absl::optional<float> speech_probability, AudioBuffer* audio);
+
+  // Handles analog level changes.
+  void NotifyAnalogLevel(int level);
+
+  static bool Validate(const AudioProcessing::Config::GainController2& config);
+
+  AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
+
+ private:
+  static std::atomic<int> instance_count_;
+  const AvailableCpuFeatures cpu_features_;
+  ApmDataDumper data_dumper_;
+  GainApplier fixed_gain_applier_;
+  std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
+  std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
+  Limiter limiter_;
+  int calls_since_last_limiter_log_;
+  int analog_level_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_