Adds delta-time logging for audio playout

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_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
+// Delta times between two successive playout callbacks are limited to this
+// value before added to an internal array.
+const size_t kMaxDeltaTimeInMs = 500;
 
 class AudioDeviceObserver;
 
 class AudioDeviceBuffer {
  public:
   AudioDeviceBuffer();
   virtual ~AudioDeviceBuffer();
 
   void SetId(uint32_t id) {};
   int32_t RegisterAudioCallback(AudioTransport* audioCallback);
@@ skipping 124 matching lines @@
   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_;
+
+  // Time stamp of last playout callback.
+  uint64_t last_playout_time_;
+
+  // An array where the position corresponds to time differences (in
+  // milliseconds) between two successive playout callbacks, and the stored
+  // value is the number of times a given time difference was found.
+  // Writing to the array is done without a lock since it is only read once at
+  // destruction when no audio is running.
+  uint32_t playout_diff_times_[kMaxDeltaTimeInMs + 1] = {0};
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_