Adds data logging in native AudioDeviceBuffer class

webrtc/modules/audio_device/audio_device_buffer.h

Index: webrtc/modules/audio_device/audio_device_buffer.h
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
index 1267e08be2cfb9696c083e95141c79ebb2917a9a..2d9f4cbe8e7ee82b0b5ceaa2d78c7f070b00fa75 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -8,9 +8,11 @@
  *  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/modules/audio_device/include/audio_device.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/typedefs.h"
@@ -63,11 +65,30 @@ class AudioDeviceBuffer {
   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.
+  // Members are only read and never modified by this method and all access is
+  // done without any locks. The stored data is only for logging purposes and
+  // minor deviations to to potential race issues are ignored.
+  // TODO(henrika): remove all usage of locks in this class, add thread checker
+  // and document the threading model.
+  void LogStats();
+
+  rtc::CriticalSection _critSect;
+  rtc::CriticalSection _critSectCb;
 
   AudioTransport* _ptrCbAudioTransport;
 
+  // Task queue posting delayed tasks periodically. Used as a timer and calls
+  // LogStats() in each task.
+  std::unique_ptr<rtc::TaskQueue> task_queue_;

[stefan-webrtc, 2016/07/07 15:22:14]

Any point in allocating this dynamically?

[henrika_webrtc, 2016/07/08 12:46:48]

Actually not. Will change.

+
+  // Ensures that the timer is only started once.
+  bool timer_has_started_;
+
   uint32_t _recSampleRate;
   uint32_t _playSampleRate;
 
@@ -107,8 +128,34 @@ class AudioDeviceBuffer {
   int _recDelayMS;
   int _clockDrift;
   int high_delay_counter_;
+
+  // 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_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_