Refactor AudioParameters member setting.

content/renderer/media/webrtc_local_audio_source_provider.cc

 // Copyright 2013 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_source_provider.h"
 
 #include "base/logging.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_fifo.h"
@@ skipping 16 matching lines @@
     : is_enabled_(false),
       track_(track),
       track_stopped_(false) {
   // Get the native audio output hardware sample-rate for the sink.
   // We need to check if RenderThreadImpl is valid here since the unittests
   // do not have one and they will inject their own |sink_params_| for testing.
   if (RenderThreadImpl::current()) {
     media::AudioHardwareConfig* hardware_config =
         RenderThreadImpl::current()->GetAudioHardwareConfig();
     int sample_rate = hardware_config->GetOutputSampleRate();
-    sink_params_.Reset(
-        media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-        media::CHANNEL_LAYOUT_STEREO, 2, sample_rate, 16,
-        kWebAudioRenderBufferSize);
+    sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+                       media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
+                       kWebAudioRenderBufferSize);
   }
 
   // Connect the source provider to the track as a sink.
   MediaStreamAudioSink::AddToAudioTrack(this, track_);
 }
 
 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
   if (audio_converter_.get())
     audio_converter_->RemoveInput(this);
 
@@ skipping 93 matching lines @@
 
   return 1.0;
 }
 
 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
     const media::AudioParameters& sink_params) {
   sink_params_ = sink_params;
 }
 
 }  // namespace content