Preparational work for an upcoming addition of a threadchecking scheme for APM

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/base/thread_checker.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);
+                  CriticalSectionWrapper* crit,
+                  const rtc::ThreadChecker* render_thread_checker,
+                  const rtc::ThreadChecker* capture_thread_checker);
   virtual ~GainControlImpl();
 
   int ProcessRenderAudio(AudioBuffer* audio);
   int AnalyzeCaptureAudio(AudioBuffer* audio);
   int ProcessCaptureAudio(AudioBuffer* audio);
 
   // ProcessingComponent implementation.
   int Initialize() override;
 
   // GainControl implementation.
@@ skipping 25 matching lines @@
   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_;
+  const rtc::ThreadChecker* const render_thread_checker_;
+  const rtc::ThreadChecker* const capture_thread_checker_;
   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_