Remove WebRtc audio source references for local audio tracks.

content/renderer/media/webrtc/processed_local_audio_source.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 
 #include "base/atomicops.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
@@ skipping 48 matching lines @@
 
   // Gets/Sets source constraints. Using this is optional, but must be done
   // before the first call to ConnectToTrack().
   const blink::WebMediaConstraints& source_constraints() const {
     return constraints_;
   }
   void SetSourceConstraints(const blink::WebMediaConstraints& constraints);
 
   // The following accessors are not valid until after the source is started
   // (when the first track is connected).
-  webrtc::AudioSourceInterface* rtc_source() const { return rtc_source_.get(); }
   const scoped_refptr<MediaStreamAudioProcessor>& audio_processor() const {
     return audio_processor_;
   }
   const scoped_refptr<MediaStreamAudioLevelCalculator::Level>& audio_level()
       const {
     return level_calculator_.level();
   }
 
   // Thread-safe volume accessors used by WebRtcAudioDeviceImpl.
   void SetVolume(int volume);
@@ skipping 40 matching lines @@
   // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output
   // data is in a unit of 10 ms data chunk.
   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
   // The device created by the AudioDeviceFactory in EnsureSourceIsStarted().
   scoped_refptr<media::AudioCapturerSource> source_;
 
   // Lock used to ensure thread-safe access to |source_| by SetVolume().
   mutable base::Lock source_lock_;
 
-  // Holder for WebRTC audio pipeline objects. Created in
-  // EnsureSourceIsStarted().
-  scoped_refptr<webrtc::AudioSourceInterface> rtc_source_;
-
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   base::subtle::Atomic32 volume_;
 
   // Used to calculate the signal level that shows in the UI.
   MediaStreamAudioLevelCalculator level_calculator_;
 
   bool allow_invalid_render_frame_id_for_testing_;
 
   DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_