Revert of Revert of Remove the last piece of deprecated synchronous IO code.

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 264 matching lines @@
 
   if (source_params_ == 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_.channels(),
-      source_params_.input_channels(), source_params_.sample_rate(),
+      source_params_.channel_layout(), source_params_.sample_rate(),
       source_params_.bits_per_sample(),
 #if defined(OS_ANDROID)
       // On Android, input and output use the same sample rate. In order to
       // use the low latency mode, we need to use the buffer size suggested by
       // the AudioManager for the sink.  It will later be used to decide
       // the buffer size of the shared memory buffer.
       frames_per_buffer_,
 #else
       2 * source_params_.frames_per_buffer(),
 #endif
@@ skipping 28 matching lines @@
     sink_->Stop();
     sink_started_ = false;
   }
 
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_,
                                               source_render_frame_id_);
   MaybeStartSink();
 }
 
 }  // namespace content