MediaStream tests need to call WebHeap::collectGarbage

content/renderer/media/rtc_peer_connection_handler_unittest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <string>
 #include <vector>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop/message_loop.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 22 matching lines @@
 #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandler.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelHandler.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
 #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandlerClient.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
 #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";
 static const char kDummySdpType[] = "dummy type";
 
 using blink::WebRTCPeerConnectionHandlerClient;
 using testing::NiceMock;
 using testing::_;
 using testing::Ref;
 
@@ skipping 159 matching lines @@
     mock_tracker_.reset(new NiceMock<MockPeerConnectionTracker>());
     blink::WebRTCConfiguration config;
     blink::WebMediaConstraints constraints;
     EXPECT_TRUE(pc_handler_->InitializeForTest(config, constraints,
                                                mock_tracker_.get()));
 
     mock_peer_connection_ = pc_handler_->native_peer_connection();
     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) {
     std::string video_track_label("video-label");
     std::string audio_track_label("audio-label");
 
     blink::WebMediaStreamSource audio_source;
     audio_source.initialize(blink::WebString::fromUTF8(audio_track_label),
                             blink::WebMediaStreamSource::TypeAudio,
                             blink::WebString::fromUTF8("audio_track"));
@@ skipping 523 matching lines @@
   scoped_refptr<webrtc::AudioTrackInterface> webrtc_track =
       remote_stream->GetAudioTracks()[0].get();
   remote_stream->RemoveTrack(webrtc_track.get());
 
   {
     blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks1;
     webkit_stream.audioTracks(modified_audio_tracks1);
     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;
   webkit_stream.audioTracks(modified_audio_tracks2);
   EXPECT_EQ(1u, modified_audio_tracks2.size());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddVideoTrackFromRemoteStream) {
   std::string remote_stream_label("remote_stream");
   scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
@@ skipping 16 matching lines @@
   // Remove the Webrtc video track from the Webrtc MediaStream.
   scoped_refptr<webrtc::VideoTrackInterface> webrtc_track =
       remote_stream->GetVideoTracks()[0].get();
   remote_stream->RemoveTrack(webrtc_track.get());
   {
     blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks1;
     webkit_stream.videoTracks(modified_video_tracks1);
     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;
   webkit_stream.videoTracks(modified_video_tracks2);
   EXPECT_EQ(1u, modified_video_tracks2.size());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, OnIceCandidate) {
   testing::InSequence sequence;
   EXPECT_CALL(*mock_tracker_.get(),
@@ skipping 47 matching lines @@
   EXPECT_CALL(*mock_tracker_.get(),
               TrackCreateDTMFSender(pc_handler_.get(),
                                     testing::Ref(tracks[0])));
 
   scoped_ptr<blink::WebRTCDTMFSenderHandler> sender(
       pc_handler_->createDTMFSender(tracks[0]));
   EXPECT_TRUE(sender.get());
 }
 
 }  // namespace content