Update webrtc&libjingle 6774:6825.

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 
 #include <vector>
 
 #include "base/memory/ref_counted.h"
@@ skipping 49 matching lines @@
       const scoped_refptr<MediaStreamAudioProcessor>& processor);
 
  private:
   // webrtc::MediaStreamTrack implementation.
   virtual std::string kind() const OVERRIDE;
 
   // webrtc::AudioTrackInterface implementation.
   virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE;
   virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE;
   virtual bool GetSignalLevel(int* level) OVERRIDE;
-  virtual talk_base::scoped_refptr<webrtc::AudioProcessorInterface>
+  virtual rtc::scoped_refptr<webrtc::AudioProcessorInterface>
       GetAudioProcessor() OVERRIDE;
 
   // cricket::AudioCapturer implementation.
   virtual void AddChannel(int channel_id) OVERRIDE;
   virtual void RemoveChannel(int channel_id) OVERRIDE;
 
   // webrtc::AudioTrackInterface implementation.
   virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE;
   virtual cricket::AudioRenderer* GetRenderer() OVERRIDE;
 
   // Weak reference.
   WebRtcLocalAudioTrack* owner_;
 
   // The source of the audio track which handles the audio constraints.
   // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
-  talk_base::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
+  rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
 
   // The audio processsor that applies audio processing on the data of audio
   // track.
   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
   // A vector of WebRtc VoE channels that the capturer sends data to.
   std::vector<int> voe_channels_;
 
   // A vector of the peer connection sink adapters which receive the audio data
   // from the audio track.
   ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
 
   // The amplitude of the signal.
   int signal_level_;
 
   // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
   mutable base::Lock lock_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_