Index: content/renderer/media/rtc_peer_connection_handler_unittest.cc |
diff --git a/content/renderer/media/rtc_peer_connection_handler_unittest.cc b/content/renderer/media/rtc_peer_connection_handler_unittest.cc |
index a79aee48b58b382a6179be4a19eaac8d0966cca4..630861d28ff6889a1e47a70487958ac33a2253a2 100644 |
--- a/content/renderer/media/rtc_peer_connection_handler_unittest.cc |
+++ b/content/renderer/media/rtc_peer_connection_handler_unittest.cc |
@@ -40,6 +40,7 @@ |
#include "third_party/WebKit/public/platform/WebRTCStatsRequest.h" |
#include "third_party/WebKit/public/platform/WebRTCVoidRequest.h" |
#include "third_party/WebKit/public/platform/WebURL.h" |
+#include "third_party/WebKit/public/web/WebHeap.h" |
#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" |
static const char kDummySdp[] = "dummy sdp"; |
@@ -219,6 +220,14 @@ class RTCPeerConnectionHandlerTest : public ::testing::Test { |
ASSERT_TRUE(mock_peer_connection_); |
} |
+ virtual void TearDown() { |
+ pc_handler_.reset(); |
+ mock_tracker_.reset(); |
+ mock_dependency_factory_.reset(); |
+ mock_client_.reset(); |
+ blink::WebHeap::collectAllGarbageForTesting(); |
+ } |
+ |
// Creates a WebKit local MediaStream. |
blink::WebMediaStream CreateLocalMediaStream( |
const std::string& stream_label) { |
@@ -762,6 +771,8 @@ TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddAudioTrackFromRemoteStream) { |
EXPECT_EQ(0u, modified_audio_tracks1.size()); |
} |
+ blink::WebHeap::collectGarbageForTesting(); |
+ |
// Add the WebRtc audio track again. |
remote_stream->AddTrack(webrtc_track.get()); |
blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks2; |
@@ -798,6 +809,8 @@ TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddVideoTrackFromRemoteStream) { |
EXPECT_EQ(0u, modified_video_tracks1.size()); |
} |
+ blink::WebHeap::collectGarbageForTesting(); |
+ |
// Add the WebRtc video track again. |
remote_stream->AddTrack(webrtc_track.get()); |
blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks2; |