webrtc/modules/audio_processing/gain_control_impl.h

/*
 *  Copyright (c) 2012 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_GAIN_CONTROL_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_

#include <vector>

#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/swap_queue.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"

namespace webrtc {

class AudioBuffer;
class CriticalSectionWrapper;

class GainControlImpl : public GainControl,
                        public ProcessingComponent {
 public:
  GainControlImpl(const AudioProcessing* apm,
                  CriticalSectionWrapper* crit);
  virtual ~GainControlImpl();

  int ProcessRenderAudio(AudioBuffer* audio);
  int AnalyzeCaptureAudio(AudioBuffer* audio);
  int ProcessCaptureAudio(AudioBuffer* audio);

  // ProcessingComponent implementation.
  int Initialize() override;

  // GainControl implementation.
  bool is_enabled() const override;
  int stream_analog_level() override;
  bool is_limiter_enabled() const override;
  Mode mode() const override;

  // Reads render side data that has been queued on the render call.
  void ReadQueuedRenderData();

 private:
  static const size_t kAllowedValuesOfSamplesPerFrame1 = 80;
  static const size_t kAllowedValuesOfSamplesPerFrame2 = 160;
  // TODO(peah): Decrease this once we properly handle hugely unbalanced
  // reverse and forward call numbers.
  static const size_t kMaxNumFramesToBuffer = 100;

  // GainControl implementation.
  int Enable(bool enable) override;
  int set_stream_analog_level(int level) override;
  int set_mode(Mode mode) override;
  int set_target_level_dbfs(int level) override;
  int target_level_dbfs() const override;
  int set_compression_gain_db(int gain) override;
  int compression_gain_db() const override;
  int enable_limiter(bool enable) override;
  int set_analog_level_limits(int minimum, int maximum) override;
  int analog_level_minimum() const override;
  int analog_level_maximum() const override;
  bool stream_is_saturated() const override;

  // ProcessingComponent implementation.
  void* CreateHandle() const override;
  int InitializeHandle(void* handle) const override;
  int ConfigureHandle(void* handle) const override;
  void DestroyHandle(void* handle) const override;
  int num_handles_required() const override;
  int GetHandleError(void* handle) const override;

  void AllocateRenderQueue();

  const AudioProcessing* apm_;
  CriticalSectionWrapper* crit_;
  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_;

  size_t render_queue_element_max_size_;
  std::vector<int16_t> render_queue_buffer_;
  std::vector<int16_t> capture_queue_buffer_;
  rtc::scoped_ptr<
      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
      render_signal_queue_;
};
}  // namespace webrtc

#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_