diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 9b137d9f76..341a73c70a 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -235,12 +235,6 @@ class AudioTrackInterface : public MediaStreamTrackInterface {
   virtual rtc::scoped_refptr<AudioProcessorInterface>
       GetAudioProcessor() { return NULL; }
 
-  // Get a pointer to the audio renderer of this AudioTrack.
-  // The pointer is valid for the lifetime of this AudioTrack.
-  // TODO(xians): Remove the following interface after Chrome switches to
-  // AddSink() and RemoveSink() interfaces.
-  virtual cricket::AudioRenderer* GetRenderer() { return NULL; }
-
  protected:
   virtual ~AudioTrackInterface() {}
 };
diff --git a/talk/app/webrtc/mediastreamtrackproxy.h b/talk/app/webrtc/mediastreamtrackproxy.h
index d97ba3c62c..b485905de7 100644
--- a/talk/app/webrtc/mediastreamtrackproxy.h
+++ b/talk/app/webrtc/mediastreamtrackproxy.h
@@ -47,7 +47,6 @@ BEGIN_PROXY_MAP(AudioTrack)
   PROXY_METHOD1(bool, GetSignalLevel, int*)
   PROXY_METHOD0(rtc::scoped_refptr<AudioProcessorInterface>,
                 GetAudioProcessor)
-  PROXY_METHOD0(cricket::AudioRenderer*, GetRenderer)
 
   PROXY_METHOD1(bool, set_enabled, bool)
   PROXY_METHOD1(bool, set_state, TrackState)
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index e37c5f84c4..a30bf0b163 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -211,10 +211,7 @@ void AudioRtpSender::SetAudioSend() {
   }
 #endif
 
-  // Use the renderer if the audio track has one, otherwise use the sink
-  // adapter owned by this class.
-  cricket::AudioRenderer* renderer =
-      track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
+  cricket::AudioRenderer* renderer = sink_adapter_.get();
   ASSERT(renderer != nullptr);
   provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
 }