Refactor AudioParameters member setting.

content/renderer/media/webrtc_audio_capturer.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_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 325 matching lines @@
   // We either default to 10ms, or use the hardware buffer size.
   if (buffer_size == 0)
     buffer_size = GetBufferSize(sample_rate);
 
   // Dispatch the new parameters both to the sink(s) and to the new source,
   // also apply the new |constraints|.
   // The idea is to get rid of any dependency of the microphone parameters
   // which would normally be used by default.
   // bits_per_sample is always 16 for now.
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                                channel_layout,
-                                sample_rate,
-                                16,
-                                buffer_size,
-                                device_info_.device.input.effects);
+                                channel_layout, sample_rate, 16, buffer_size);
+  params.set_effects(device_info_.device.input.effects);
 
   {
     base::AutoLock auto_lock(lock_);
     // Notify the |audio_processor_| of the new format.
     audio_processor_->OnCaptureFormatChanged(params);
 
     // Notify all tracks about the new format.
     tracks_.TagAll();
   }
 
@@ skipping 258 matching lines @@
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSourceInternal(source,
                             params.channel_layout(),
                             params.sample_rate(),
                             0);
 }
 
 }  // namespace content