Adds delta-time logging for audio playout

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 ee6b2297c77e8e3a981615e52d3c0ccf0db7c767..f4be649285b93001045ef5517176dd6e829f13ab 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
 
+#include <map>

[magjed_webrtc, 2016/07/29 13:13:20]

remove

+
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_checker.h"
@@ -23,6 +25,9 @@ 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;
 
@@ -167,6 +172,16 @@ class AudioDeviceBuffer {
 
   // 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