Refactor AudioParameters member setting.

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/location.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/synchronization/lock.h"
@@ skipping 244 matching lines @@
 
   DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()";
 
   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(),
-      WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
-                                                frames_per_buffer_),
-      source_params_.effects());
-
+  sink_params_ = source_params_;
+  sink_params_.set_frames_per_buffer(WebRtcAudioRenderer::GetOptimalBufferSize(
+      source_params_.sample_rate(), frames_per_buffer_));
   {
     // Note: The max buffer is fairly large, but will rarely be used.
     // Cast needs the buffer to hold at least one second of audio.
     // The clock accuracy is set to 20ms because clock accuracy is
     // ~15ms on windows.
     media::AudioShifter* const new_shifter = new media::AudioShifter(
         base::TimeDelta::FromSeconds(2),
         base::TimeDelta::FromMilliseconds(20),
         base::TimeDelta::FromSeconds(20),
         source_params_.sample_rate(),
@@ skipping 11 matching lines @@
   if (sink_started_) {
     sink_->Stop();
     sink_started_ = false;
   }
 
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
   MaybeStartSink();
 }
 
 }  // namespace content