Refactoring AudioParameters |operator==| to |Equals()| method

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 255 matching lines @@
       RenderFrameImpl::FromRoutingID(source_render_frame_id_);
   MediaStreamDispatcher* const dispatcher = frame ?
       frame->GetMediaStreamDispatcher() : NULL;
   if (dispatcher && dispatcher->IsAudioDuckingActive()) {
     DVLOG(1) << "Forcing DUCKING to be ON for output";
     implicit_ducking_effect = media::AudioParameters::DUCKING;
   } else {
     DVLOG(1) << "DUCKING not forced ON for output";
   }
 
-  if (source_params_ == params)
+  if (source_params_.Equals(params))
     return;
 
   // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
   // the new format.
 
   source_params_ = params;
 
   sink_params_ = media::AudioParameters(source_params_.format(),
       source_params_.channel_layout(), source_params_.sample_rate(),
       source_params_.bits_per_sample(),
@@ skipping 37 matching lines @@
     sink_->Stop();
     sink_started_ = false;
   }
 
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_,
                                               source_render_frame_id_);
   MaybeStartSink();
 }
 
 }  // namespace content