Refactor AudioParameters member setting.

content/renderer/media/webaudio_capturer_source.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/webaudio_capturer_source.h"
 
 #include "base/logging.h"
 #include "base/time/time.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 
@@ skipping 31 matching lines @@
     return;
   }
 
   ChannelLayout channel_layout =
       number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
 
   base::AutoLock auto_lock(lock_);
   // Set the format used by this WebAudioCapturerSource. We are using 10ms data
   // as buffer size since that is the native buffer size of WebRtc packet
   // running on.
-  params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                channel_layout, number_of_channels, sample_rate, 16,
-                sample_rate / 100);
+  params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
+                sample_rate, 16, sample_rate / 100);
   audio_format_changed_ = true;
 
   wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
   capture_bus_ = AudioBus::Create(params_);
   fifo_.reset(new AudioFifo(
       params_.channels(),
       kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
 }
 
 void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
@@ skipping 66 matching lines @@
 // WebAudioCapturerSource reference still registered as an audio consumer on
 // the blink side.
 void WebAudioCapturerSource::removeFromBlinkSource() {
   if (!blink_source_.isNull()) {
     blink_source_.removeAudioConsumer(this);
     blink_source_.reset();
   }
 }
 
 }  // namespace content