Lock scheme #5: Applied the render queueing to the agc

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_