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

webrtc/modules/audio_processing/audio_processing_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_AUDIO_PROCESSING_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
 #include <list>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 
 class AgcManagerDirect;
 class AudioBuffer;
 class AudioConverter;
 
 template<typename T>
 class Beamformer;
@@ skipping 85 matching lines @@
   NoiseSuppression* noise_suppression() const override;
   VoiceDetection* voice_detection() const override;
 
  protected:
   // Overridden in a mock.
   virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
  private:
   int InitializeLocked(const ProcessingConfig& config)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  int MaybeInitializeLockedRender(const ProcessingConfig& config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  int MaybeInitializeLockedCapture(const ProcessingConfig& config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
   int MaybeInitializeLocked(const ProcessingConfig& config)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
   // TODO(ekm): Remove once all clients updated to new interface.
   int AnalyzeReverseStream(const float* const* src,
                            const StreamConfig& input_config,
                            const StreamConfig& output_config);
   int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   bool is_data_processed() const;
   bool output_copy_needed(bool is_data_processed) const;
   bool synthesis_needed(bool is_data_processed) const;
   bool analysis_needed(bool is_data_processed) const;
   bool is_rev_processed() const;
   bool rev_conversion_needed() const;
+  // TODO(peah): Add EXCLUSIVE_LOCKS_REQUIRED for the method below.
+  bool render_check_rev_conversion_needed() const;
   void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   EchoCancellationImpl* echo_cancellation_;
   EchoControlMobileImpl* echo_control_mobile_;
   GainControlImpl* gain_control_;
   HighPassFilterImpl* high_pass_filter_;
   LevelEstimatorImpl* level_estimator_;
   NoiseSuppressionImpl* noise_suppression_;
   VoiceDetectionImpl* voice_detection_;
   rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
 
   std::list<ProcessingComponent*> component_list_;
   CriticalSectionWrapper* crit_;
+  rtc::ThreadChecker render_thread_checker_;
+  rtc::ThreadChecker capture_thread_checker_;
+  rtc::ThreadChecker signal_thread_checker_;
   rtc::scoped_ptr<AudioBuffer> render_audio_;
   rtc::scoped_ptr<AudioBuffer> capture_audio_;
   rtc::scoped_ptr<AudioConverter> render_converter_;
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // TODO(andrew): make this more graceful. Ideally we would split this stuff
   // out into a separate class with an "enabled" and "disabled" implementation.
   int WriteMessageToDebugFile();
   int WriteInitMessage();
 
   // Writes Config message. If not |forced|, only writes the current config if
@@ skipping 52 matching lines @@
   const std::vector<Point> array_geometry_;
   const SphericalPointf target_direction_;
 
   bool intelligibility_enabled_;
   rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_