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

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/buffer.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/task_queue.h"

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

can you fwd declare TaskQueue now instead of including here?

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

Done.

 #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 {
 // Delta times between two successive playout callbacks are limited to this
 // value before added to an internal array.
 const size_t kMaxDeltaTimeInMs = 500;
 // TODO(henrika): remove when no longer used by external client.
 const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
 
 class AudioDeviceObserver;
 
 class AudioDeviceBuffer {
  public:
   AudioDeviceBuffer();
   virtual ~AudioDeviceBuffer();
 
   void SetId(uint32_t id) {};
   int32_t RegisterAudioCallback(AudioTransport* audio_callback);
 
   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);
   int32_t RecordingSampleRate() const;
   int32_t PlayoutSampleRate() const;
 
   int32_t SetRecordingChannels(size_t channels);
   int32_t SetPlayoutChannels(size_t channels);
   size_t RecordingChannels() const;
   size_t PlayoutChannels() const;
@@ skipping 18 matching lines @@
   int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopOutputFileRecording();
 
   int32_t SetTypingStatus(bool typing_status);
 
  private:
   // Posts the first delayed task in the task queue and starts the periodic
   // 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();
 
   // Clears all members tracking stats for recording and playout.
   void ResetRecStats();
   void ResetPlayStats();
 
   // 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(int16_t max_abs, size_t num_samples);
   void UpdatePlayStats(int16_t max_abs, size_t num_samples);
 
   // Ensures that methods are called on the same thread as the thread that
   // creates this object.
   rtc::ThreadChecker thread_checker_;
 
   // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
   // and it must outlive this object.
   AudioTransport* audio_transport_cb_;
 
   // TODO(henrika): given usage of thread checker, it should be possible to
   // remove all locks in this class.
   rtc::CriticalSection lock_;
   rtc::CriticalSection lock_cb_;
 
   // 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_;
   uint32_t play_sample_rate_;
 
   // Number of audio channels.
   size_t rec_channels_;
   size_t play_channels_;
 
   // Number of bytes per audio sample (2 or 4).
@@ skipping 67 matching lines @@
   // Contains max level (max(abs(x))) of recorded audio packets over the last
   // 10 seconds where a new measurement is done twice per second. The level
   // is reset to zero at each call to LogStats(). Only modified on the task
   // queue thread.
   int16_t max_rec_level_;
 
   // Contains max level of recorded audio packets over the last 10 seconds
   // 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
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_