Refactor AudioParameters member setting.

content/renderer/media/webrtc_local_audio_source_provider_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/logging.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 
 namespace content {
 
 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  protected:
   void SetUp() override {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                         media::CHANNEL_LAYOUT_MONO, 1, 48000, 16, 480);
+                         media::CHANNEL_LAYOUT_MONO, 48000, 16, 480);
     sink_params_.Reset(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-        media::CHANNEL_LAYOUT_STEREO, 2, 44100, 16,
+        media::CHANNEL_LAYOUT_STEREO, 44100, 16,
         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
     sink_bus_ = media::AudioBus::Create(sink_params_);
     MockMediaConstraintFactory constraint_factory;
     scoped_refptr<WebRtcAudioCapturer> capturer(
         WebRtcAudioCapturer::CreateCapturer(
             -1, StreamDeviceInfo(),
             constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
     scoped_ptr<WebRtcLocalAudioTrack> native_track(
@@ skipping 28 matching lines @@
   // Point the WebVector into memory owned by |sink_bus_|.
   blink::WebVector<float*> audio_data(
       static_cast<size_t>(sink_bus_->channels()));
   for (size_t i = 0; i < audio_data.size(); ++i)
     audio_data[i] = sink_bus_->channel(i);
 
   // Enable the |source_provider_| by asking for data. This will inject
   // source_params_.frames_per_buffer() of zero into the resampler since there
   // no available data in the FIFO.
   source_provider_->provideInput(audio_data, sink_params_.frames_per_buffer());
-  EXPECT_TRUE(sink_bus_->channel(0)[0] == 0);
+  EXPECT_EQ(0, sink_bus_->channel(0)[0]);
 
   // Create a source AudioBus with channel data filled with non-zero values.
   const scoped_ptr<media::AudioBus> source_bus =
       media::AudioBus::Create(source_params_);
   std::fill(source_bus->channel(0),
             source_bus->channel(0) + source_bus->frames(),
             0.5f);
 
   // Deliver data to |source_provider_|.
   base::TimeTicks estimated_capture_time = base::TimeTicks::Now();
@@ skipping 47 matching lines @@
   // Stop the audio track.
   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
       MediaStreamTrack::GetTrack(blink_track_));
   native_track->Stop();
 
   // Delete the source provider.
   source_provider_.reset();
 }
 
 }  // namespace content