Remove unused RenderIO() interface.

content/renderer/media/webrtc_local_audio_renderer.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_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/metrics/histogram.h"
@@ skipping 269 matching lines @@
 
   base::AutoLock auto_lock(thread_lock_);
 
   // Clear up the old data in the FIFO.
   loopback_fifo_->Clear();
 
   if (!sink_params_.IsValid() || !playing_ || !volume_ || sink_started_)
     return;
 
   DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_.";
-  sink_->InitializeUnifiedStream(sink_params_, this, session_id_);
+  sink_->InitializeWithSessionId(sink_params_, this, session_id_);
   sink_->Start();
   sink_started_ = true;
   UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
                             kSinkStarted, kSinkStatesMax);
 }
 
 void WebRtcLocalAudioRenderer::ReconfigureSink(
     const media::AudioParameters& params) {
   DCHECK(message_loop_->BelongsToCurrentThread());
 
   DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()";
 
   if (!sink_)
     return;  // WebRtcLocalAudioRenderer has not yet been started.
 
   // Stop |sink_| and re-create a new one to be initialized with different audio
   // parameters.  Then, invoke MaybeStartSink() to restart everything again.
   if (sink_started_) {
     sink_->Stop();
     sink_started_ = false;
   }
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_,
                                               source_render_frame_id_);
   MaybeStartSink();
 }
 
 }  // namespace content