Chromium Code Reviews| 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..0bbd7a1260cfd143bfd0461998efa62e82bbc198 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(); |
|
haraken
2014/09/09 04:51:16
The reason we need to reset blink_source_ here is
|
| + blink::WebHeap::collectAllGarbage(); |
| + } |
| + |
| media::AudioParameters params_; |
| blink::WebMediaStreamSource blink_source_; |
| scoped_refptr<MockCapturerSource> capturer_source_; |