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

content/renderer/media/webrtc_audio_capturer.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 "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 246 matching lines @@
   DVLOG(1) << "WebRtcAudioCapturer::AddTrack()";
 
   {
     base::AutoLock auto_lock(lock_);
     // Verify that |track| is not already added to the list.
     DCHECK(!tracks_.Contains(TrackOwner::TrackWrapper(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());
   }
 }
 
 void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::RemoveTrack()";
   bool stop_source = false;
   {
     base::AutoLock auto_lock(lock_);
 
@@ skipping 102 matching lines @@
   // WebRtc native buffer size.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
                     input_params.channel_layout(),
                     static_cast<float>(input_params.sample_rate()));
 }
 
 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.
   // We need to set the AGC control before starting the stream.
   source_->SetAutomaticGainControl(true);
   source_->Start();
   running_ = true;
 }
 
 void WebRtcAudioCapturer::Stop() {
@@ skipping 146 matching lines @@
     }
   }
 }
 
 void WebRtcAudioCapturer::OnCaptureError() {
   NOTIMPLEMENTED();
 }
 
 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(
     int* session_id,
     int* output_sample_rate,
     int* output_frames_per_buffer) const {
   // Don't set output parameters unless all of them are valid.
   if (device_info_.session_id <= 0 ||
       !device_info_.device.matched_output.sample_rate ||
       !device_info_.device.matched_output.frames_per_buffer)
@@ skipping 40 matching lines @@
 
 void WebRtcAudioCapturer::SetCapturerSourceForTesting(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSource(source, params.channel_layout(),
                     static_cast<float>(params.sample_rate()));
 }
 
 }  // namespace content