Refactor AudioParameters member setting.

content/renderer/media/webrtc_local_audio_track_unittest.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 "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
@@ skipping 95 matching lines @@
   void Start() override {
     audio_thread_.reset(new FakeAudioThread(capturer_, params_));
     audio_thread_->Start();
     OnStart();
   }
   void Stop() override {
     audio_thread_->Stop();
     audio_thread_.reset();
     OnStop();
   }
+
  protected:
   ~MockCapturerSource() override {}
 
  private:
   scoped_ptr<FakeAudioThread> audio_thread_;
   WebRtcAudioCapturer* capturer_;
   media::AudioParameters params_;
 };
 
 class MockMediaStreamAudioSink : public MediaStreamAudioSink {
@@ skipping 19 matching lines @@
  private:
   media::AudioParameters params_;
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   void SetUp() override {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                  media::CHANNEL_LAYOUT_STEREO, 2, 48000, 16, 480);
+                  media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480);
     MockMediaConstraintFactory constraint_factory;
     blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
                              "dummy",
                              false /* remote */, true /* readonly */);
     MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
     blink_source_.setExtraData(audio_source);
 
     StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
                             std::string(), std::string());
     capturer_ = WebRtcAudioCapturer::CreateCapturer(
@@ skipping 333 matching lines @@
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source.get(), OnStop()).Times(1);
   capturer->Stop();
 
   // Even though this test don't use |capturer_source_| it will be stopped
   // during teardown of the test harness.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
 }
 
 }  // namespace content