1 files changed, 80 insertions, 0 deletions
diff --git a/src/modules/audio_processing/gain_control_impl.h b/src/modules/audio_processing/gain_control_impl.h
new file mode 100644
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+++ b/src/modules/audio_processing/gain_control_impl.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 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_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
+
+#include <vector>
+
+#include "audio_processing.h"
+#include "processing_component.h"
+
+namespace webrtc {
+class AudioProcessingImpl;
+class AudioBuffer;
+
+class GainControlImpl : public GainControl,
+                        public ProcessingComponent {
+ public:
+  explicit GainControlImpl(const AudioProcessingImpl* apm);
+  virtual ~GainControlImpl();
+
+  int ProcessRenderAudio(AudioBuffer* audio);
+  int AnalyzeCaptureAudio(AudioBuffer* audio);
+  int ProcessCaptureAudio(AudioBuffer* audio);
+
+  // ProcessingComponent implementation.
+  virtual int Initialize();
+  virtual int get_version(char* version, int version_len_bytes) const;
+
+  // GainControl implementation.
+  virtual bool is_enabled() const;
+  virtual int stream_analog_level();
+
+ private:
+  // GainControl implementation.
+  virtual int Enable(bool enable);
+  virtual int set_stream_analog_level(int level);
+  virtual int set_mode(Mode mode);
+  virtual Mode mode() const;
+  virtual int set_target_level_dbfs(int level);
+  virtual int target_level_dbfs() const;
+  virtual int set_compression_gain_db(int gain);
+  virtual int compression_gain_db() const;
+  virtual int enable_limiter(bool enable);
+  virtual bool is_limiter_enabled() const;
+  virtual int set_analog_level_limits(int minimum, int maximum);
+  virtual int analog_level_minimum() const;
+  virtual int analog_level_maximum() const;
+  virtual bool stream_is_saturated() const;
+
+  // ProcessingComponent implementation.
+  virtual void* CreateHandle() const;
+  virtual int InitializeHandle(void* handle) const;
+  virtual int ConfigureHandle(void* handle) const;
+  virtual int DestroyHandle(void* handle) const;
+  virtual int num_handles_required() const;
+  virtual int GetHandleError(void* handle) const;
+
+  const AudioProcessingImpl* apm_;
+  Mode mode_;
+  int minimum_capture_level_;
+  int maximum_capture_level_;
+  bool limiter_enabled_;
+  int target_level_dbfs_;
+  int compression_gain_db_;
+  std::vector<int> capture_levels_;
+  int analog_capture_level_;
+  bool was_analog_level_set_;
+  bool stream_is_saturated_;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_