Fix data race in WebRtcAudioRenderer.

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 467cbe35f74daf8e6459575f096488f78c166f62..c41b69fd56269fc9483a76a0a243c99fe83ae048 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -302,15 +302,6 @@ void WebRtcAudioRenderer::EnterPauseState() {
 void WebRtcAudioRenderer::Stop() {
   DVLOG(1) << "WebRtcAudioRenderer::Stop()";
   DCHECK(thread_checker_.CalledOnValidThread());
-  // If |max_render_time_| is zero, no render call has been made.
-  if (!max_render_time_.is_zero()) {
-    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)
@@ -326,6 +317,19 @@ void WebRtcAudioRenderer::Stop() {
     state_ = UNINITIALIZED;
   }
 
+  // Apart from here, |max_render_time_| is only accessed in SourceCallback(),
+  // which is guaranteed to not run after |source_| has been set to null, and
+  // not before this function has returned.
+  // If |max_render_time_| is zero, no render call has been made.
+  if (!max_render_time_.is_zero()) {
+    UMA_HISTOGRAM_CUSTOM_COUNTS(
+        "Media.Audio.Render.GetSourceDataTimeMax.WebRTC",
+        max_render_time_.InMicroseconds(),
+        kRenderTimeHistogramMinMicroseconds,
+        kRenderTimeHistogramMaxMicroseconds, 50);
+    max_render_time_ = base::TimeDelta();
+  }
+
   // Make sure to stop the sink while _not_ holding the lock since the Render()
   // callback may currently be executing and trying to grab the lock while we're
   // stopping the thread on which it runs.