MediaStream content_unittests need to trigger a GC before tear down

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

 // Copyright 2013 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 "base/logging.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
+#include "third_party/WebKit/public/web/WebHeap.h"
 
 namespace content {
 
 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                          media::CHANNEL_LAYOUT_MONO, 1, 48000, 16, 480);
     sink_params_.Reset(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
@@ skipping 17 matching lines @@
                             blink::WebMediaStreamSource::TypeAudio,
                             base::UTF8ToUTF16("dummy_source_name"));
     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
                             audio_source);
     blink_track_.setExtraData(native_track.release());
     source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
     source_provider_->SetSinkParamsForTesting(sink_params_);
     source_provider_->OnSetFormat(source_params_);
   }
 
+  virtual void TearDown() OVERRIDE {
+    source_provider_.reset();
+    blink_track_.reset();
+    blink::WebHeap::collectAllGarbageForTesting();
+  }
+
   media::AudioParameters source_params_;
   scoped_ptr<int16[]> source_data_;
   media::AudioParameters sink_params_;
   scoped_ptr<media::AudioBus> sink_bus_;
   blink::WebMediaStreamTrack blink_track_;
   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
 };
 
 TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
   // Point the WebVector into memory owned by |sink_bus_|.
@@ skipping 66 matching lines @@
   // Stop the audio track.
   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
       MediaStreamTrack::GetTrack(blink_track_));
   native_track->Stop();
 
   // Delete the source provider.
   source_provider_.reset();
 }
 
 }  // namespace content