MediaStream tests need to call WebHeap::collectGarbage

content/renderer/media/webrtc_local_audio_track_unittest.cc

Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 20c462e15d3804f4c2b39bd99f9fe0b7f3971a9d..d3d074bcd6a6d6bfaed602469e800a3995402561 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -16,6 +16,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
+#include "third_party/WebKit/public/web/WebHeap.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 using ::testing::_;
@@ -190,6 +191,11 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
   }
 
+  virtual void TearDown() OVERRIDE {
+    blink_source_.reset();
+    blink::WebHeap::collectAllGarbageForTesting();
+  }
+
   media::AudioParameters params_;
   blink::WebMediaStreamSource blink_source_;
   scoped_refptr<MockCapturerSource> capturer_source_;