Remove implicit conversions from scoped_refptr to T* in content/renderer/media/webrtc*

This patch was generated by running the rewrite_scoped_refptr clang tool
on a Linux build.

BUG=110610

Review URL: https://codereview.chromium.org/503683003

Cr-Commit-Position: refs/heads/master@{#292096}

content/renderer/media/webrtc/media_stream_remote_video_source.cc

@@ -151,21 +151,21 @@ void MediaStreamRemoteVideoSource::StartSourceImpl(
     const media::VideoCaptureParams& params,
     const VideoCaptureDeliverFrameCB& frame_callback) {
   DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!delegate_);
+  DCHECK(!delegate_.get());
   delegate_ = new RemoteVideoSourceDelegate(io_message_loop(), frame_callback);
-  remote_track_->AddRenderer(delegate_);
+  remote_track_->AddRenderer(delegate_.get());
   OnStartDone(MEDIA_DEVICE_OK);
 }
 
 void MediaStreamRemoteVideoSource::StopSourceImpl() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(state() != MediaStreamVideoSource::ENDED);
-  remote_track_->RemoveRenderer(delegate_);
+  remote_track_->RemoveRenderer(delegate_.get());
 }
 
 webrtc::VideoRendererInterface*
 MediaStreamRemoteVideoSource::RenderInterfaceForTest() {
-  return delegate_;
+  return delegate_.get();
 }
 
 void MediaStreamRemoteVideoSource::OnChanged() {

content/renderer/media/webrtc/media_stream_remote_video_source_unittest.cc

@@ -35,12 +35,11 @@ class MediaStreamRemoteVideoSourceTest
   MediaStreamRemoteVideoSourceTest()
       : child_process_(new ChildProcess()),
         mock_factory_(new MockPeerConnectionDependencyFactory()),
-        webrtc_video_track_(
-            mock_factory_->CreateLocalVideoTrack(
-                "test",
-                static_cast<cricket::VideoCapturer*>(NULL))),
-        remote_source_(
-            new MediaStreamRemoteVideoSourceUnderTest(webrtc_video_track_)),
+        webrtc_video_track_(mock_factory_->CreateLocalVideoTrack(
+            "test",
+            static_cast<cricket::VideoCapturer*>(NULL))),
+        remote_source_(new MediaStreamRemoteVideoSourceUnderTest(
+            webrtc_video_track_.get())),
         number_of_successful_constraints_applied_(0),
         number_of_failed_constraints_applied_(0) {
     webkit_source_.initialize(base::UTF8ToUTF16("dummy_source_id"),

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

@@ -180,7 +180,7 @@ PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
 
 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
   CleanupPeerConnectionFactory();
-  if (aec_dump_message_filter_)
+  if (aec_dump_message_filter_.get())
     aec_dump_message_filter_->RemoveDelegate(this);
 }
 
@@ -224,7 +224,7 @@ bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
     // be called multiple times which is likely also a bug.
     return false;
   }
-  source_data->SetAudioCapturer(capturer);
+  source_data->SetAudioCapturer(capturer.get());
 
   // Creates a LocalAudioSource object which holds audio options.
   // TODO(xians): The option should apply to the track instead of the source.
@@ -237,7 +237,7 @@ bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
     return false;
   }
-  source_data->SetLocalAudioSource(rtc_source);
+  source_data->SetLocalAudioSource(rtc_source.get());
   return true;
 }
 
@@ -248,7 +248,7 @@ PeerConnectionDependencyFactory::CreateVideoCapturer(
   // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
   // since the base class of WebRtcVideoCapturerAdapter is a
   // cricket::VideoCapturer and it uses the libjingle thread wrappers.
-  if (!GetPcFactory())
+  if (!GetPcFactory().get())
     return NULL;
   return new WebRtcVideoCapturerAdapter(is_screeencast);
 }
@@ -265,9 +265,9 @@ PeerConnectionDependencyFactory::CreateVideoSource(
 
 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
 PeerConnectionDependencyFactory::GetPcFactory() {
-  if (!pc_factory_)
+  if (!pc_factory_.get())
     CreatePeerConnectionFactory();
-  CHECK(pc_factory_);
+  CHECK(pc_factory_.get());
   return pc_factory_;
 }
 
@@ -326,12 +326,12 @@ void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
   scoped_refptr<media::GpuVideoAcceleratorFactories> gpu_factories =
       RenderThreadImpl::current()->GetGpuFactories();
   if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) {
-    if (gpu_factories)
+    if (gpu_factories.get())
       decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
   }
 
   if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) {
-    if (gpu_factories)
+    if (gpu_factories.get())
       encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
   }
 
@@ -348,7 +348,7 @@ void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
                                           audio_device_.get(),
                                           encoder_factory.release(),
                                           decoder_factory.release()));
-  CHECK(factory);
+  CHECK(factory.get());
 
   pc_factory_ = factory;
   webrtc::PeerConnectionFactoryInterface::Options factory_options;
@@ -364,7 +364,7 @@ void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
     // In unit tests not creating a message filter, |aec_dump_message_filter_|
     // will be NULL. We can just ignore that. Other unit tests and browser tests
     // ensure that we do get the filter when we should.
-    if (aec_dump_message_filter_)
+    if (aec_dump_message_filter_.get())
       aec_dump_message_filter_->AddDelegate(this);
   }
 }
@@ -381,7 +381,7 @@ PeerConnectionDependencyFactory::CreatePeerConnection(
     webrtc::PeerConnectionObserver* observer) {
   CHECK(web_frame);
   CHECK(observer);
-  if (!GetPcFactory())
+  if (!GetPcFactory().get())
     return NULL;
 
   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
@@ -449,10 +449,8 @@ void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
   // TODO(xians): Merge |source| to the capturer(). We can't do this today
   // because only one capturer() is supported while one |source| is created
   // for each audio track.
-  scoped_ptr<WebRtcLocalAudioTrack> audio_track(
-      new WebRtcLocalAudioTrack(adapter,
-                                source_data->GetAudioCapturer(),
-                                webaudio_source));
+  scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
+      adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
 
   StartLocalAudioTrack(audio_track.get());
 
@@ -656,7 +654,7 @@ void PeerConnectionDependencyFactory::OnIpcClosing() {
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
-  if (audio_device_)
+  if (audio_device_.get())
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();

content/renderer/media/webrtc/video_destination_handler.cc

@@ -97,7 +97,7 @@ void PpFrameWriter::StartSourceImpl(
     const media::VideoCaptureParams& params,
     const VideoCaptureDeliverFrameCB& frame_callback) {
   DCHECK(CalledOnValidThread());
-  DCHECK(!delegate_);
+  DCHECK(!delegate_.get());
   DVLOG(3) << "PpFrameWriter::StartSourceImpl()";
   delegate_ = new FrameWriterDelegate(io_message_loop(), frame_callback);
   OnStartDone(MEDIA_DEVICE_OK);

content/renderer/media/webrtc/video_destination_handler_unittest.cc

@@ -94,7 +94,7 @@ TEST_F(VideoDestinationHandlerTest, PutFrame) {
 
     EXPECT_CALL(sink, OnVideoFrame()).WillOnce(
         RunClosure(quit_closure));
-    frame_writer->PutFrame(image, 10);
+    frame_writer->PutFrame(image.get(), 10);
     run_loop.Run();
   }
   // TODO(perkj): Verify that the track output I420 when

content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

@@ -41,7 +41,7 @@ class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
     capturer_ = WebRtcAudioCapturer::CreateCapturer(
         -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
         constraint_factory.CreateWebMediaConstraints(), NULL, NULL);
-    track_.reset(new WebRtcLocalAudioTrack(adapter_, capturer_, NULL));
+    track_.reset(new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL));
   }
 
  protected:

content/renderer/media/webrtc/webrtc_media_stream_adapter.cc

@@ -64,7 +64,7 @@ void WebRtcMediaStreamAdapter::TrackRemoved(
 
     for (ScopedVector<WebRtcVideoTrackAdapter>::iterator it =
              video_adapters_.begin(); it != video_adapters_.end(); ++it) {
-      if ((*it)->webrtc_video_track() == webrtc_track) {
+      if ((*it)->webrtc_video_track() == webrtc_track.get()) {
         video_adapters_.erase(it);
         break;
       }
@@ -93,7 +93,7 @@ void WebRtcMediaStreamAdapter::CreateAudioTrack(
   const blink::WebMediaStreamSource& source = track.source();
   MediaStreamAudioSource* audio_source =
       static_cast<MediaStreamAudioSource*>(source.extraData());
-  if (audio_source && audio_source->GetAudioCapturer())
+  if (audio_source && audio_source->GetAudioCapturer().get())
     audio_source->GetAudioCapturer()->EnablePeerConnectionMode();
 
   webrtc_media_stream_->AddTrack(native_track->GetAudioAdapter());

content/renderer/media/webrtc_audio_capturer.cc

@@ -264,7 +264,7 @@ void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) {
     // Add with a tag, so we remember to call OnSetFormat() on the new
     // track.
     scoped_refptr<TrackOwner> track_owner(new TrackOwner(track));
-    tracks_.AddAndTag(track_owner);
+    tracks_.AddAndTag(track_owner.get());
   }
 }
 
@@ -387,7 +387,7 @@ void WebRtcAudioCapturer::Start() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Start()";
   base::AutoLock auto_lock(lock_);
-  if (running_ || !source_)
+  if (running_ || !source_.get())
     return;
 
   // Start the data source, i.e., start capturing data from the current source.
@@ -554,8 +554,8 @@ void WebRtcAudioCapturer::OnCaptureError() {
 
 media::AudioParameters WebRtcAudioCapturer::source_audio_parameters() const {
   base::AutoLock auto_lock(lock_);
-  return audio_processor_ ?
-      audio_processor_->InputFormat() : media::AudioParameters();
+  return audio_processor_.get() ? audio_processor_->InputFormat()
+                                : media::AudioParameters();
 }
 
 bool WebRtcAudioCapturer::GetPairedOutputParameters(

content/renderer/media/webrtc_audio_capturer_unittest.cc

@@ -105,7 +105,7 @@ class WebRtcAudioCapturerTest : public testing::Test {
 
     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-    track_.reset(new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
+    track_.reset(new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
     track_->Start();
 
     // Connect a mock sink to the track.
@@ -124,7 +124,8 @@ class WebRtcAudioCapturerTest : public testing::Test {
     audio_bus->Zero();
 
     media::AudioCapturerSource::CaptureCallback* callback =
-        static_cast<media::AudioCapturerSource::CaptureCallback*>(capturer_);
+        static_cast<media::AudioCapturerSource::CaptureCallback*>(
+            capturer_.get());
 
     // Verify the sink is getting the correct values.
     EXPECT_CALL(*sink, FormatIsSet());
@@ -185,7 +186,7 @@ TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) {
                                params_.frames_per_buffer()),
           constraint_factory.CreateWebMediaConstraints(), NULL, NULL)
   );
-  EXPECT_TRUE(capturer == NULL);
+  EXPECT_TRUE(capturer.get() == NULL);
 }
 
 

content/renderer/media/webrtc_audio_device_impl.cc

@@ -202,7 +202,7 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
 
 void WebRtcAudioDeviceImpl::RemoveAudioRenderer(WebRtcAudioRenderer* renderer) {
   DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK_EQ(renderer, renderer_);
+  DCHECK_EQ(renderer, renderer_.get());
   base::AutoLock auto_lock(lock_);
   // Notify the playout sink of the change.
   for (PlayoutDataSinkList::const_iterator it = playout_sinks_.begin();
@@ -401,7 +401,7 @@ int32_t WebRtcAudioDeviceImpl::MinMicrophoneVolume(uint32_t* min_volume) const {
 
 int32_t WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable(bool* available) const {
   DCHECK(initialized_);
-  *available = renderer_ && renderer_->channels() == 2;
+  *available = renderer_.get() && renderer_->channels() == 2;
   return 0;
 }
 
@@ -444,7 +444,7 @@ int32_t WebRtcAudioDeviceImpl::RecordingSampleRate(
 
 int32_t WebRtcAudioDeviceImpl::PlayoutSampleRate(
     uint32_t* sample_rate) const {
-  *sample_rate = renderer_ ? renderer_->sample_rate() : 0;
+  *sample_rate = renderer_.get() ? renderer_->sample_rate() : 0;
   return 0;
 }
 

content/renderer/media/webrtc_audio_renderer_unittest.cc

@@ -91,7 +91,7 @@ class WebRtcAudioRendererTest : public testing::Test {
         renderer_(new WebRtcAudioRenderer(stream_, 1, 1, 1, 44100, 441)) {
     EXPECT_CALL(*factory_.get(), CreateOutputDevice(1))
         .WillOnce(Return(mock_output_device_));
-    EXPECT_CALL(*mock_output_device_, Start());
+    EXPECT_CALL(*mock_output_device_.get(), Start());
     EXPECT_TRUE(renderer_->Initialize(source_.get()));
     renderer_proxy_ = renderer_->CreateSharedAudioRendererProxy(stream_);
   }
@@ -114,7 +114,7 @@ TEST_F(WebRtcAudioRendererTest, StopRenderer) {
 
   // |renderer_| has only one proxy, stopping the proxy should stop the sink of
   // |renderer_|.
-  EXPECT_CALL(*mock_output_device_, Stop());
+  EXPECT_CALL(*mock_output_device_.get(), Stop());
   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
   renderer_proxy_->Stop();
 }
@@ -136,16 +136,16 @@ TEST_F(WebRtcAudioRendererTest, MultipleRenderers) {
 
   // Stop the |renderer_proxy_| should not stop the sink since it is used by
   // other proxies.
-  EXPECT_CALL(*mock_output_device_, Stop()).Times(0);
+  EXPECT_CALL(*mock_output_device_.get(), Stop()).Times(0);
   renderer_proxy_->Stop();
 
   for (int i = 0; i < kNumberOfRendererProxy; ++i) {
     if (i != kNumberOfRendererProxy -1) {
-      EXPECT_CALL(*mock_output_device_, Stop()).Times(0);
+      EXPECT_CALL(*mock_output_device_.get(), Stop()).Times(0);
     } else {
       // When the last proxy is stopped, the sink will stop.
       EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
-      EXPECT_CALL(*mock_output_device_, Stop());
+      EXPECT_CALL(*mock_output_device_.get(), Stop());
     }
     renderer_proxies_[i]->Stop();
   }

content/renderer/media/webrtc_local_audio_renderer.cc

@@ -153,7 +153,7 @@ void WebRtcLocalAudioRenderer::Stop() {
   // Stop the output audio stream, i.e, stop asking for data to render.
   // It is safer to call Stop() on the |sink_| to clean up the resources even
   // when the |sink_| is never started.
-  if (sink_) {
+  if (sink_.get()) {
     sink_->Stop();
     sink_ = NULL;
   }
@@ -316,7 +316,7 @@ void WebRtcLocalAudioRenderer::ReconfigureSink(
     loopback_fifo_.reset(new_fifo);
   }
 
-  if (!sink_)
+  if (!sink_.get())
     return;  // WebRtcLocalAudioRenderer has not yet been started.
 
   // Stop |sink_| and re-create a new one to be initialized with different audio

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

@@ -37,7 +37,7 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
     scoped_ptr<WebRtcLocalAudioTrack> native_track(
-        new WebRtcLocalAudioTrack(adapter, capturer, NULL));
+        new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
     blink::WebMediaStreamSource audio_source;
     audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
                             blink::WebMediaStreamSource::TypeAudio,

content/renderer/media/webrtc_local_audio_track.cc

@@ -86,7 +86,7 @@ void WebRtcLocalAudioTrack::Capture(const int16* audio_data,
                                    volume,
                                    need_audio_processing,
                                    key_pressed);
-    if (new_volume != 0 && capturer.get() && !webaudio_source_) {
+    if (new_volume != 0 && capturer.get() && !webaudio_source_.get()) {
       // Feed the new volume to WebRtc while changing the volume on the
       // browser.
       capturer->SetVolume(new_volume);
@@ -135,7 +135,7 @@ void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
   // we remember to call OnSetFormat() on the new sink.
   scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
       new MediaStreamAudioSinkOwner(sink));
-  sinks_.AddAndTag(sink_owner);
+  sinks_.AddAndTag(sink_owner.get());
 }
 
 void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
@@ -169,7 +169,7 @@ void WebRtcLocalAudioTrack::AddSink(PeerConnectionAudioSink* sink) {
   // we remember to call OnSetFormat() on the new sink.
   scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
       new PeerConnectionAudioSinkOwner(sink));
-  sinks_.AddAndTag(sink_owner);
+  sinks_.AddAndTag(sink_owner.get());
 }
 
 void WebRtcLocalAudioTrack::RemoveSink(PeerConnectionAudioSink* sink) {

content/renderer/media/webrtc_local_audio_track_unittest.cc

@@ -181,7 +181,7 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
     capturer_ = WebRtcAudioCapturer::CreateCapturer(
         -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
         audio_source);
-    audio_source->SetAudioCapturer(capturer_);
+    audio_source->SetAudioCapturer(capturer_.get());
     capturer_source_ = new MockCapturerSource(capturer_.get());
     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
         .WillOnce(Return());
@@ -204,7 +204,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
   track->Start();
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 
@@ -238,7 +238,7 @@ TEST_F(WebRtcLocalAudioTrackTest,  DISABLED_DisableEnableAudioTrack) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
   track->Start();
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
@@ -272,7 +272,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
-    new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
   track_1->Start();
   EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
@@ -290,7 +290,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
-    new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
   track_2->Start();
   EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
 
@@ -329,7 +329,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
   track->Start();
 
   // When the track goes away, it will automatically stop the
@@ -345,13 +345,13 @@ TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track1(
-      new WebRtcLocalAudioTrack(adapter1, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL));
   track1->Start();
 
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track2(
-      new WebRtcLocalAudioTrack(adapter2, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL));
   track2->Start();
 
   track1->Stop();
@@ -368,7 +368,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
   track_1->Start();
 
   // Verify the data flow by connecting the sink to |track_1|.
@@ -386,7 +386,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
   track_2->Start();
 
   // Stop the capturer will clear up the track lists in the capturer.
@@ -418,7 +418,7 @@ TEST_F(WebRtcLocalAudioTrackTest,
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
-      new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
+      new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
   track_1->Start();
 
   // Verify the data flow by connecting the |sink_1| to |track_1|.
@@ -451,7 +451,7 @@ TEST_F(WebRtcLocalAudioTrackTest,
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
-      new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL));
+      new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
   track_2->Start();
 
   // Verify the data flow by connecting the |sink_2| to |track_2|.
@@ -505,7 +505,7 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
-      new WebRtcLocalAudioTrack(adapter, capturer, NULL));
+      new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
   track->Start();
 
   // Verify the data flow by connecting the |sink| to |track|.
@@ -526,7 +526,7 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
   EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
 
   // Stopping the new source will stop the second track.
-  EXPECT_CALL(*source, OnStop()).Times(1);
+  EXPECT_CALL(*source.get(), OnStop()).Times(1);
   capturer->Stop();
 
   // Even though this test don't use |capturer_source_| it will be stopped