Improvements in how WebRTC.Audio.RecordedOnlyZeros is added as histogram

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 6967ebd757147762b5291abd25d8c1a7940e0b0e..4841ffdd0c710211391d4b2fe5cbc6b89bd00da5 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -38,6 +38,10 @@ class AudioDeviceBuffer {
 
   int32_t InitPlayout();
   int32_t InitRecording();
+  void StartPlayout();

[tommi, 2016/10/25 16:43:23]

we have the following three states:

construct / destruct
init / (no uninitialize function afaict)
start / stop

Do we need all of them?

[henrika_webrtc, 2016/10/26 12:42:41]

Good point. Calls to init have been around for a long time. Will see if I can
remove them without touching too much code...

It worked well and I was able to remove both Init methods since they serve no
purpose.

+  void StartRecording();
+  void StopPlayout();
+  void StopRecording();
 
   int32_t SetRecordingSampleRate(uint32_t fsHz);
   int32_t SetPlayoutSampleRate(uint32_t fsHz);
@@ -76,6 +80,9 @@ class AudioDeviceBuffer {
   // timer.
   void StartTimer();
 
+  // Stops the timer by releasing the unique TaskQueue instance.
+  void StopTimer();
+
   // Called periodically on the internal thread created by the TaskQueue.
   void LogStats();
 
@@ -105,10 +112,13 @@ class AudioDeviceBuffer {
   // 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_;
+  std::unique_ptr<rtc::TaskQueue> task_queue_;
 
-  // Ensures that the timer is only started once.
-  bool timer_has_started_;
+  // Keeps track of if playout/recording are active or not. A combination
+  // of these states are used to determine when to start and stop the timer.
+  // Only used on the creating thread and not used to control any media flow.
+  bool playing_;
+  bool recording_;
 
   // Sample rate in Hertz.
   uint32_t rec_sample_rate_;
@@ -196,18 +206,19 @@ class AudioDeviceBuffer {
   // where a new measurement is done twice per second.
   int16_t max_play_level_;
 
-  // Counts number of times we detect "no audio" corresponding to a case where
-  // all level measurements since the last log has been exactly zero.
-  // In other words: this counter is incremented only if 20 measurements
-  // (two per second) in a row equals zero. The member is only incremented on
-  // the task queue and max once every 10th second.
-  size_t num_rec_level_is_zero_;
-
   // Counts number of audio callbacks modulo 50 to create a signal when
   // a new storage of audio stats shall be done.
   // Only updated on the OS-specific audio thread that drives audio.
   int16_t rec_stat_count_;
   int16_t play_stat_count_;
+
+  // Time stamps of when playout and recording starts.
+  uint64_t play_start_time_;
+  uint64_t rec_start_time_;
+
+  // Set to true at construction and modified to false as soon as one audio-
+  // level estimate larger than zero is detected.
+  bool only_silence_recorded_;
 };
 
 }  // namespace webrtc