Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat

webrtc/modules/audio_device/ios/audio_device_ios.mm

Index: webrtc/modules/audio_device/ios/audio_device_ios.mm
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index fd2b2e7422780d50ae024e5a5185370f4c46cdff..b224359307f5ceb78ba6fc0afad57439f29739f2 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -245,6 +245,7 @@ int32_t AudioDeviceIOS::StartPlayout() {
   }
   rtc::AtomicOps::ReleaseStore(&playing_, 1);
   num_playout_callbacks_ = 0;
+  num_detected_playout_glitches_ = 0;
   return 0;
 }
 
@@ -263,7 +264,7 @@ int32_t AudioDeviceIOS::StopPlayout() {
   // 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;
-  if (num_detected_playout_glitches_ > 0) {
+  if (num_playout_callbacks_ > 0 && num_detected_playout_glitches_ > 0) {

[minyue-webrtc, 2017/07/05 08:20:21]

num_playout_callbacks_ > 0 is redundant if, by design,
num_detected_playout_glitches_  <= num_playout_callbacks_

I'd add 
RTC_DCHECK_GE(num_playout_callbacks_, num_detected_playout_glitches_).

[henrika_webrtc, 2017/07/05 08:39:35]

Acknowledged.

     average_number_of_playout_callbacks_between_glitches =
         num_playout_callbacks_ / num_detected_playout_glitches_;
   }