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

content/renderer/media/webrtc_audio_device_impl.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 "content/renderer/media/webrtc_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 184 matching lines @@
   // Pass the render data to the playout sinks.
   base::AutoLock auto_lock(lock_);
   for (PlayoutDataSinkList::const_iterator it = playout_sinks_.begin();
        it != playout_sinks_.end(); ++it) {
     (*it)->OnPlayoutData(audio_bus, sample_rate, audio_delay_milliseconds);
   }
 }
 
 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();
        it != playout_sinks_.end(); ++it) {
     (*it)->OnPlayoutDataSourceChanged();
   }
 
   renderer_ = NULL;
   playing_ = false;
 }
@@ skipping 178 matching lines @@
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::MinMicrophoneVolume(uint32_t* min_volume) const {
   *min_volume = 0;
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable(bool* available) const {
   DCHECK(initialized_);
-  *available = renderer_ && renderer_->channels() == 2;
+  *available = renderer_.get() && renderer_->channels() == 2;
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::StereoRecordingIsAvailable(
     bool* available) const {
   DCHECK(initialized_);
   // TODO(xians): These kind of hardware methods do not make much sense since we
   // support multiple sources. Remove or figure out new APIs for such methods.
   scoped_refptr<WebRtcAudioCapturer> capturer(GetDefaultCapturer());
   if (!capturer.get())
@@ skipping 22 matching lines @@
   if (!capturer.get())
     return -1;
 
   *sample_rate = static_cast<uint32_t>(
       capturer->source_audio_parameters().sample_rate());
   return 0;
 }
 
 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;
 }
 
 bool WebRtcAudioDeviceImpl::SetAudioRenderer(WebRtcAudioRenderer* renderer) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(renderer);
 
   base::AutoLock auto_lock(lock_);
   if (renderer_.get())
     return false;
@@ skipping 62 matching lines @@
   // If there is no capturer or there are more than one open capture devices,
   // return false.
   if (capturers_.empty() || capturers_.size() > 1)
     return false;
 
   return GetDefaultCapturer()->GetPairedOutputParameters(
       session_id, output_sample_rate, output_frames_per_buffer);
 }
 
 }  // namespace content