Ensures that AudioDeviceBuffer::StopPeriodicLogging works as intended

webrtc/modules/audio_device/audio_device_buffer.cc

 /*
  *  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.
  */
 
@@ skipping 50 matching lines @@
       last_rec_samples_(0),
       play_samples_(0),
       last_play_samples_(0),
       max_rec_level_(0),
       max_play_level_(0),
       last_timer_task_time_(0),
       rec_stat_count_(0),
       play_stat_count_(0),
       play_start_time_(0),
       rec_start_time_(0),
-      only_silence_recorded_(true) {
+      only_silence_recorded_(true),
+      log_stats_(false) {
   LOG(INFO) << "AudioDeviceBuffer::ctor";
   playout_thread_checker_.DetachFromThread();
   recording_thread_checker_.DetachFromThread();
 }
 
 AudioDeviceBuffer::~AudioDeviceBuffer() {
   RTC_DCHECK_RUN_ON(&main_thread_checker_);
   RTC_DCHECK(!playing_);
   RTC_DCHECK(!recording_);
   LOG(INFO) << "AudioDeviceBuffer::~dtor";
@@ skipping 340 matching lines @@
 }
 
 void AudioDeviceBuffer::StopPeriodicLogging() {
   task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
                                  AudioDeviceBuffer::LOG_STOP));
 }
 
 void AudioDeviceBuffer::LogStats(LogState state) {
   RTC_DCHECK_RUN_ON(&task_queue_);
   int64_t now_time = rtc::TimeMillis();
+
   if (state == AudioDeviceBuffer::LOG_START) {
     // Reset counters at start. We will not add any logging in this state but
     // the timer will started by posting a new (delayed) task.
     num_stat_reports_ = 0;
     last_timer_task_time_ = now_time;
+    log_stats_ = true;
   } else if (state == AudioDeviceBuffer::LOG_STOP) {
     // Stop logging and posting new tasks.
+    log_stats_ = false;
+  } else if (state == AudioDeviceBuffer::LOG_ACTIVE) {
+    // Keep logging unless logging was disabled while task was posted.
+  }
+
+  // Avoid adding more logs since we are in STOP mode.
+  if (!log_stats_) {
     return;
-  } else if (state == AudioDeviceBuffer::LOG_ACTIVE) {
-    // Default state. Just keep on logging.
   }
 
   int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds;
   int64_t time_since_last = rtc::TimeDiff(now_time, last_timer_task_time_);
   last_timer_task_time_ = now_time;
 
   // Log the latest statistics but skip the first round just after state was
   // set to LOG_START. Hence, first printed log will be after ~10 seconds.
   if (++num_stat_reports_ > 1 && time_since_last > 0) {
     uint32_t diff_samples = rec_samples_ - last_rec_samples_;
@@ skipping 65 matching lines @@
                                         size_t samples_per_channel) {
   RTC_DCHECK_RUN_ON(&task_queue_);
   ++play_callbacks_;
   play_samples_ += samples_per_channel;
   if (max_abs > max_play_level_) {
     max_play_level_ = max_abs;
   }
 }
 
 }  // namespace webrtc