Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  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 227 matching lines @@
   if (!recording_ &&
       audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
     if (!audio_unit_->Start()) {
       RTCLogError(@"StartPlayout failed to start audio unit.");
       return -1;
     }
     LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
   }
   rtc::AtomicOps::ReleaseStore(&playing_, 1);
   num_playout_callbacks_ = 0;
+  num_detected_playout_glitches_ = 0;
   return 0;
 }
 
 int32_t AudioDeviceIOS::StopPlayout() {
   LOGI() << "StopPlayout";
   RTC_DCHECK_RUN_ON(&thread_checker_);
   if (!audio_is_initialized_ || !playing_) {
     return 0;
   }
   if (!recording_) {
     ShutdownPlayOrRecord();
     audio_is_initialized_ = false;
   }
   rtc::AtomicOps::ReleaseStore(&playing_, 0);
 
   // Derive average number of calls to OnGetPlayoutData() between detected
   // audio glitches and add the result to a histogram.
   int average_number_of_playout_callbacks_between_glitches = 100000;
+  RTC_DCHECK_GE(num_playout_callbacks_, num_detected_playout_glitches_);
   if (num_detected_playout_glitches_ > 0) {
     average_number_of_playout_callbacks_between_glitches =
         num_playout_callbacks_ / num_detected_playout_glitches_;
   }
   RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches",
       average_number_of_playout_callbacks_between_glitches);
   RTCLog(@"Average number of playout callbacks between glitches: %d",
          average_number_of_playout_callbacks_between_glitches);
   return 0;
@@ skipping 662 matching lines @@
 
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
   [session lockForConfiguration];
   UnconfigureAudioSession();
   [session endWebRTCSession:nil];
   [session unlockForConfiguration];
 }
 
 }  // namespace webrtc