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

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 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 "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 162 matching lines @@
     P2PSocketDispatcher* p2p_socket_dispatcher)
     : network_manager_(NULL),
       p2p_socket_dispatcher_(p2p_socket_dispatcher),
       signaling_thread_(NULL),
       worker_thread_(NULL),
       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
 }
 
 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
   CleanupPeerConnectionFactory();
-  if (aec_dump_message_filter_)
+  if (aec_dump_message_filter_.get())
     aec_dump_message_filter_->RemoveDelegate(this);
 }
 
 blink::WebRTCPeerConnectionHandler*
 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
     blink::WebRTCPeerConnectionHandlerClient* client) {
   // Save histogram data so we can see how much PeerConnetion is used.
   // The histogram counts the number of calls to the JS API
   // webKitRTCPeerConnection.
   UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
@@ skipping 23 matching lines @@
                           source_data));
   if (!capturer.get()) {
     DLOG(WARNING) << "Failed to create the capturer for device "
         << device_info.device.id;
     // TODO(xians): Don't we need to check if source_observer is observing
     // something? If not, then it looks like we have a leak here.
     // OTOH, if it _is_ observing something, then the callback might
     // 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.
   // TODO(perkj): Move audio constraints parsing to Chrome.
   // Currently there are a few constraints that are parsed by libjingle and
   // the state is set to ended if parsing fails.
   scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
       CreateLocalAudioSource(&constraints).get());
   if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
     return false;
   }
-  source_data->SetLocalAudioSource(rtc_source);
+  source_data->SetLocalAudioSource(rtc_source.get());
   return true;
 }
 
 WebRtcVideoCapturerAdapter*
 PeerConnectionDependencyFactory::CreateVideoCapturer(
     bool is_screeencast) {
   // We need to make sure the libjingle thread wrappers have been created
   // 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);
 }
 
 scoped_refptr<webrtc::VideoSourceInterface>
 PeerConnectionDependencyFactory::CreateVideoSource(
     cricket::VideoCapturer* capturer,
     const blink::WebMediaConstraints& constraints) {
   RTCMediaConstraints webrtc_constraints(constraints);
   scoped_refptr<webrtc::VideoSourceInterface> source =
       GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
   return source;
 }
 
 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_;
 }
 
 void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
   DCHECK(!pc_factory_.get());
   DCHECK(!signaling_thread_);
   DCHECK(!worker_thread_);
   DCHECK(!network_manager_);
   DCHECK(!socket_factory_);
   DCHECK(!chrome_worker_thread_.IsRunning());
@@ skipping 38 matching lines @@
   net::EnsureNSSSSLInit();
 #endif
 
   scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
   scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
 
   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
   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));
   }
 
 #if defined(OS_ANDROID)
   if (!media::MediaCodecBridge::SupportsSetParameters())
     encoder_factory.reset();
 #endif
 
   EnsureWebRtcAudioDeviceImpl();
 
   scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
       webrtc::CreatePeerConnectionFactory(worker_thread_,
                                           signaling_thread_,
                                           audio_device_.get(),
                                           encoder_factory.release(),
                                           decoder_factory.release()));
-  CHECK(factory);
+  CHECK(factory.get());
 
   pc_factory_ = factory;
   webrtc::PeerConnectionFactoryInterface::Options factory_options;
   factory_options.disable_sctp_data_channels = false;
   factory_options.disable_encryption =
       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
   pc_factory_->SetOptions(factory_options);
 
   // TODO(xians): Remove the following code after kDisableAudioTrackProcessing
   // is removed.
   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) {
     aec_dump_message_filter_ = AecDumpMessageFilter::Get();
     // 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);
   }
 }
 
 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
   return pc_factory_.get() != NULL;
 }
 
 scoped_refptr<webrtc::PeerConnectionInterface>
 PeerConnectionDependencyFactory::CreatePeerConnection(
     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
     const webrtc::MediaConstraintsInterface* constraints,
     blink::WebFrame* web_frame,
     webrtc::PeerConnectionObserver* observer) {
   CHECK(web_frame);
   CHECK(observer);
-  if (!GetPcFactory())
+  if (!GetPcFactory().get())
     return NULL;
 
   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
         new rtc::RefCountedObject<P2PPortAllocatorFactory>(
             p2p_socket_dispatcher_.get(),
             network_manager_,
             socket_factory_.get(),
             web_frame);
 
   PeerConnectionIdentityService* identity_service =
@@ skipping 47 matching lines @@
   // Creates an adapter to hold all the libjingle objects.
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
                                            source_data->local_audio_source()));
   static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
       track.isEnabled());
 
   // 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());
 
   // Pass the ownership of the native local audio track to the blink track.
   blink::WebMediaStreamTrack writable_track = track;
   writable_track.setExtraData(audio_track.release());
 }
 
 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
     WebRtcLocalAudioTrack* audio_track) {
@@ skipping 183 matching lines @@
   DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
   // Do nothing. We never disable AEC dump for non-track-processing case.
 }
 
 void PeerConnectionDependencyFactory::OnIpcClosing() {
   DCHECK(CalledOnValidThread());
   aec_dump_message_filter_ = NULL;
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
-  if (audio_device_)
+  if (audio_device_.get())
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content