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

content/renderer/media/webrtc_local_audio_track.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_local_audio_track.h"
 
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/media_stream_audio_level_calculator.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_sink_owner.h"
@@ skipping 68 matching lines @@
        ++it) {
     int new_volume = (*it)->OnData(audio_data,
                                    audio_parameters_.sample_rate(),
                                    audio_parameters_.channels(),
                                    audio_parameters_.frames_per_buffer(),
                                    voe_channels,
                                    delay.InMilliseconds(),
                                    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);
     }
   }
 }
 
 void WebRtcLocalAudioTrack::OnSetFormat(
     const media::AudioParameters& params) {
   DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
@@ skipping 28 matching lines @@
   // Verify that |sink| is not already added to the list.
   DCHECK(!sinks_.Contains(
       MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)));
 
   // Create (and add to the list) a new MediaStreamAudioTrackSink
   // which owns the |sink| and delagates all calls to the
   // MediaStreamAudioSink interface. It will be tagged in the list, so
   // 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) {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
 
   base::AutoLock auto_lock(lock_);
 
   scoped_refptr<MediaStreamAudioTrackSink> removed_item = sinks_.Remove(
       MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
@@ skipping 13 matching lines @@
   // Verify that |sink| is not already added to the list.
   DCHECK(!sinks_.Contains(
       MediaStreamAudioTrackSink::WrapsPeerConnectionSink(sink)));
 
   // Create (and add to the list) a new MediaStreamAudioTrackSink
   // which owns the |sink| and delagates all calls to the
   // MediaStreamAudioSink interface. It will be tagged in the list, so
   // 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) {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
 
   base::AutoLock auto_lock(lock_);
 
   scoped_refptr<MediaStreamAudioTrackSink> removed_item = sinks_.Remove(
       MediaStreamAudioTrackSink::WrapsPeerConnectionSink(sink));
@@ skipping 59 matching lines @@
 
   for (SinkList::ItemList::const_iterator it = sinks.begin();
        it != sinks.end();
        ++it){
     (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
     (*it)->Reset();
   }
 }
 
 }  // namespace content