Lock scheme #8: Introduced the new locking scheme

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/criticalsection.h"
 #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;
@@ skipping 64 matching lines @@
   int AnalyzeReverseStream(const float* const* data,
                            size_t samples_per_channel,
                            int sample_rate_hz,
                            ChannelLayout layout) override;
   int ProcessReverseStream(const float* const* src,
                            const StreamConfig& reverse_input_config,
                            const StreamConfig& reverse_output_config,
                            float* const* dest) override;
   int set_stream_delay_ms(int delay) override;
   int stream_delay_ms() const override;
-  bool was_stream_delay_set() const override;
+  bool was_stream_delay_set() const override
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
   int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
   int StartDebugRecording(FILE* handle) override;
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
   void UpdateHistogramsOnCallEnd() override;
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
   HighPassFilter* high_pass_filter() const override;
   LevelEstimator* level_estimator() const override;
   NoiseSuppression* noise_suppression() const override;
   VoiceDetection* voice_detection() const override;
 
  protected:
   // Overridden in a mock.
-  virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  virtual int InitializeLocked()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
  private:
   int InitializeLocked(const ProcessingConfig& config)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int MaybeInitializeLocked(const ProcessingConfig& config)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  int MaybeInitialize(const ProcessingConfig& config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
   // 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_);
+  int AnalyzeReverseStreamLocked(const float* const* src,
+                                 const StreamConfig& input_config,
+                                 const StreamConfig& output_config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
 
-  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;
-  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_);
+  bool is_data_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool output_copy_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool synthesis_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool analysis_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool is_rev_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  bool rev_conversion_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  void InitializeExperimentalAgc()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeTransient()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeBeamformer()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeIntelligibility()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   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_;
+  std::list<ProcessingComponent*> component_list_ GUARDED_BY(crit_capture_);
+  mutable rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  mutable rtc::CriticalSection crit_capture_;
   rtc::ThreadChecker render_thread_;
   rtc::ThreadChecker capture_thread_;
-  rtc::scoped_ptr<AudioBuffer> render_audio_;
-  rtc::scoped_ptr<AudioBuffer> capture_audio_;
-  rtc::scoped_ptr<AudioConverter> render_converter_;
+  rtc::scoped_ptr<AudioBuffer> render_audio_ GUARDED_BY(crit_render_);
+  rtc::scoped_ptr<AudioBuffer> capture_audio_ GUARDED_BY(crit_capture_);
+  rtc::scoped_ptr<AudioConverter> render_converter_ GUARDED_BY(crit_render_);
 #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
   // it is different from the last saved one; if |forced|, writes the config
   // regardless of the last saved.
-  int WriteConfigMessage(bool forced);
+  int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   rtc::scoped_ptr<FileWrapper> debug_file_;
   rtc::scoped_ptr<audioproc::Event> event_msg_;  // Protobuf message.
   std::string event_str_;  // Memory for protobuf serialization.
 
   // Serialized string of last saved APM configuration.
   std::string last_serialized_config_;
 #endif
 
   // State that is written to while holding both the render and capture locks
   // but can be read while holding only one of the locks.
   struct SharedState {
     SharedState()
         :  // Format of processing streams at input/output call sites.
           api_format_({{{kSampleRate16kHz, 1, false},
                         {kSampleRate16kHz, 1, false},
                         {kSampleRate16kHz, 1, false},
                         {kSampleRate16kHz, 1, false}}}) {}
     ProcessingConfig api_format_;
   } shared_state_;
 
   // Only the rate and samples fields of fwd_proc_format_ are used because the
   // forward processing number of channels is mutable and is tracked by the
   // capture_audio_.
   StreamConfig fwd_proc_format_;
   StreamConfig rev_proc_format_;
   int split_rate_;
 
-  int stream_delay_ms_;
-  int delay_offset_ms_;
-  bool was_stream_delay_set_;
-  int last_stream_delay_ms_;
-  int last_aec_system_delay_ms_;
-  int stream_delay_jumps_;
-  int aec_system_delay_jumps_;
+  int stream_delay_ms_ GUARDED_BY(crit_capture_);
+  int delay_offset_ms_ GUARDED_BY(crit_capture_);
+  bool was_stream_delay_set_ GUARDED_BY(crit_capture_);
+  int last_stream_delay_ms_ GUARDED_BY(crit_capture_);
+  int last_aec_system_delay_ms_ GUARDED_BY(crit_capture_);
+  int stream_delay_jumps_ GUARDED_BY(crit_capture_);
+  int aec_system_delay_jumps_ GUARDED_BY(crit_capture_);
 
-  bool output_will_be_muted_ GUARDED_BY(crit_);
+  bool output_will_be_muted_ GUARDED_BY(crit_capture_);
 
-  bool key_pressed_;
+  bool key_pressed_ GUARDED_BY(crit_capture_);
 
   // Only set through the constructor's Config parameter.
   const bool use_new_agc_;
-  rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
-  int agc_startup_min_volume_;
+  rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_capture_);
+  int agc_startup_min_volume_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
 
-  bool transient_suppressor_enabled_;
-  rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
-  const bool beamformer_enabled_;
-  rtc::scoped_ptr<Beamformer<float>> beamformer_;
-  const std::vector<Point> array_geometry_;
+  bool transient_suppressor_enabled_ GUARDED_BY(crit_capture_);
+
+  rtc::scoped_ptr<TransientSuppressor> transient_suppressor_
+      GUARDED_BY(crit_capture_);
+  const bool beamformer_enabled_ GUARDED_BY(crit_capture_);
+  rtc::scoped_ptr<Beamformer<float>> beamformer_ GUARDED_BY(crit_capture_);
+  const std::vector<Point> array_geometry_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
 
   bool intelligibility_enabled_;
   rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_