Add UMA histograms for time spent getting WebRTC audio render data.

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index e3ce2b0f6775112015b1586bed60f1a90d0f97d2..414499d8e37bed2a07a9ccd7b1144452284446fa 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -34,11 +34,6 @@ namespace content {
 
 namespace {
 
-// We add a UMA histogram measuring the execution time of the Render() method
-// every |kNumCallbacksBetweenRenderTimeHistograms| callback. Assuming 10ms
-// between each callback leads to one UMA update each 100ms.
-const int kNumCallbacksBetweenRenderTimeHistograms = 10;
-
 // Audio parameters that don't change.
 const media::AudioParameters::Format kFormat =
     media::AudioParameters::AUDIO_PCM_LOW_LATENCY;
@@ -46,6 +41,10 @@ const media::ChannelLayout kChannelLayout = media::CHANNEL_LAYOUT_STEREO;
 const int kChannels = 2;
 const int kBitsPerSample = 16;
 
+// Used for UMA histograms.
+const int kRenderTimeHistogramMinMicroseconds = 100;
+const int kRenderTimeHistogramMaxMicroseconds = 30 * 1000;  // 30 milliseconds
+
 // This is a simple wrapper class that's handed out to users of a shared
 // WebRtcAudioRenderer instance.  This class maintains the per-user 'playing'
 // and 'started' states to avoid problems related to incorrect usage which
@@ -172,7 +171,7 @@ WebRtcAudioRenderer::WebRtcAudioRenderer(
       sink_params_(kFormat, kChannelLayout, 0, kBitsPerSample, 0),
       output_device_id_(device_id),
       security_origin_(security_origin),
-      render_callback_count_(0) {
+      source_render_called_(false) {
   WebRtcLogMessage(base::StringPrintf(
       "WAR::WAR. source_render_frame_id=%d, session_id=%d, effects=%i",
       source_render_frame_id, session_id, sink_params_.effects()));
@@ -251,7 +250,7 @@ void WebRtcAudioRenderer::Play() {
     return;
 
   playing_state_.set_playing(true);
-  render_callback_count_ = 0;
+  source_render_called_ = false;
 
   OnPlayStateChanged(media_stream_, &playing_state_);
 }
@@ -305,6 +304,14 @@ void WebRtcAudioRenderer::EnterPauseState() {
 void WebRtcAudioRenderer::Stop() {
   DVLOG(1) << "WebRtcAudioRenderer::Stop()";
   DCHECK(thread_checker_.CalledOnValidThread());
+  if (source_render_called_) {

[o1ka, 2016/08/24 11:52:42]

No need for the flag: you can just check (!max_render_time_.is_zero());
or (max_render_time_ != base::TimeDelta).

[Henrik Grunell, 2016/08/24 12:16:09]

Of course. We can safely assume that the render call takes non-zero time and use
that. :) Done.

+    UMA_HISTOGRAM_CUSTOM_COUNTS(
+        "Media.Audio.Render.GetSourceDataTimeMax.WebRTC",
+        max_render_time_.InMicroseconds(), kRenderTimeHistogramMinMicroseconds,
+        kRenderTimeHistogramMaxMicroseconds, 50);
+    max_render_time_ = base::TimeDelta();
+  }
+
   {
     base::AutoLock auto_lock(lock_);
     if (state_ == UNINITIALIZED)
@@ -476,11 +483,17 @@ void WebRtcAudioRenderer::SourceCallback(
   if (state_ != PLAYING)
     audio_bus->Zero();
 
-  if (++render_callback_count_ == kNumCallbacksBetweenRenderTimeHistograms) {
-    base::TimeDelta elapsed = base::TimeTicks::Now() - start_time;
-    render_callback_count_ = 0;
-    UMA_HISTOGRAM_TIMES("WebRTC.AudioRenderTimes", elapsed);
-  }
+  // Measure the elapsed time for this function and log it.
+  base::TimeDelta elapsed = base::TimeTicks::Now() - start_time;
+  UMA_HISTOGRAM_CUSTOM_COUNTS("Media.Audio.Render.GetSourceDataTime.WebRTC",
+                              elapsed.InMicroseconds(),
+                              kRenderTimeHistogramMinMicroseconds,
+                              kRenderTimeHistogramMaxMicroseconds, 50);
+
+  // Store max render call time.
+  if (elapsed > max_render_time_)
+    max_render_time_ = elapsed;
+  source_render_called_ = true;
 }
 
 void WebRtcAudioRenderer::UpdateSourceVolume(