Reland of Adds data logging in native AudioDeviceBuffer class

webrtc/modules/audio_device/audio_device_buffer.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_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
 
 const uint32_t kPulsePeriodMs = 1000;
 const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
 
@@ skipping 32 matching lines @@
   virtual int32_t GetPlayoutData(void* audioBuffer);
 
   int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopInputFileRecording();
   int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopOutputFileRecording();
 
   int32_t SetTypingStatus(bool typingStatus);
 
  private:
-  CriticalSectionWrapper& _critSect;
-  CriticalSectionWrapper& _critSectCb;
+  // Posts the first delayed task in the task queue and starts the periodic
+  // timer.
+  void StartTimer();
+
+  // Called periodically on the internal thread created by the TaskQueue.
+  void LogStats();
+
+  // Updates counters in each play/record callback but does it on the task
+  // queue to ensure that they can be read by LogStats() without any locks since
+  // each task is serialized by the task queue.
+  void UpdateRecStats(size_t num_samples);
+  void UpdatePlayStats(size_t num_samples);
+
+  // Ensures that methods are called on the same thread as the thread that
+  // creates this object.
+  rtc::ThreadChecker thread_checker_;
+
+  rtc::CriticalSection _critSect;
+  rtc::CriticalSection _critSectCb;
 
   AudioTransport* _ptrCbAudioTransport;
 
+  // Task queue used to invoke LogStats() periodically. Tasks are executed on a
+  // worker thread but it does not necessarily have to be the same thread for
+  // each task.
+  rtc::TaskQueue task_queue_;
+
+  // Ensures that the timer is only started once.
+  bool timer_has_started_;
+
   uint32_t _recSampleRate;
   uint32_t _playSampleRate;
 
   size_t _recChannels;
   size_t _playChannels;
 
   // selected recording channel (left/right/both)
   AudioDeviceModule::ChannelType _recChannel;
 
   // 2 or 4 depending on mono or stereo
@@ skipping 19 matching lines @@
 
   uint32_t _currentMicLevel;
   uint32_t _newMicLevel;
 
   bool _typingStatus;
 
   int _playDelayMS;
   int _recDelayMS;
   int _clockDrift;
   int high_delay_counter_;
+
+  // Counts number of times LogStats() has been called.
+  size_t num_stat_reports_;
+
+  // Total number of recording callbacks where the source provides 10ms audio
+  // data each time.
+  uint64_t rec_callbacks_;
+
+  // Total number of recording callbacks stored at the last timer task.
+  uint64_t last_rec_callbacks_;
+
+  // Total number of playback callbacks where the sink asks for 10ms audio
+  // data each time.
+  uint64_t play_callbacks_;
+
+  // Total number of playout callbacks stored at the last timer task.
+  uint64_t last_play_callbacks_;
+
+  // Total number of recorded audio samples.
+  uint64_t rec_samples_;
+
+  // Total number of recorded samples stored at the previous timer task.
+  uint64_t last_rec_samples_;
+
+  // Total number of played audio samples.
+  uint64_t play_samples_;
+
+  // Total number of played samples stored at the previous timer task.
+  uint64_t last_play_samples_;
+
+  // Time stamp of last stat report.
+  uint64_t last_log_stat_time_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_