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

content/renderer/media/webrtc_audio_capturer_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/command_line.h"
 #include "base/logging.h"
 #include "content/public/common/content_switches.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"
@@ skipping 87 matching lines @@
                              params_.frames_per_buffer()),
         constraints, NULL, NULL);
     capturer_source_ = new MockCapturerSource();
     EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1));
     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
     EXPECT_CALL(*capturer_source_.get(), Start());
     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
 
     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.
     scoped_ptr<MockPeerConnectionAudioSink> sink(
         new MockPeerConnectionAudioSink());
     track_->AddSink(sink.get());
 
     int delay_ms = 65;
     bool key_pressed = true;
     double volume = 0.9;
 
     // MaxVolume() in WebRtcAudioCapturer is hard-coded to return 255, we add
     // 0.5 to do the correct truncation like the production code does.
     int expected_volume_value = volume * capturer_->MaxVolume() + 0.5;
     scoped_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(params_);
     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());
     EXPECT_CALL(*sink,
                 OnDataCallback(_, _, delay_ms, expected_volume_value,
                                need_audio_processing, key_pressed))
         .Times(AtLeast(1));
     callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed);
 
     // Verify the cached values in the capturer fits what we expect.
@@ skipping 40 matching lines @@
   constraint_factory.AddMandatory(dummy_constraint, true);
 
   scoped_refptr<WebRtcAudioCapturer> capturer(
       WebRtcAudioCapturer::CreateCapturer(
           0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
                                "", "", params_.sample_rate(),
                                params_.channel_layout(),
                                params_.frames_per_buffer()),
           constraint_factory.CreateWebMediaConstraints(), NULL, NULL)
   );
-  EXPECT_TRUE(capturer == NULL);
+  EXPECT_TRUE(capturer.get() == NULL);
 }
 
 
 }  // namespace content