Revert of Revert of Remove the last piece of deprecated synchronous IO code.

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"
 
 namespace content {
 
 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                         media::CHANNEL_LAYOUT_MONO, 1, 0, 48000, 16, 480);
+                         media::CHANNEL_LAYOUT_MONO, 1, 48000, 16, 480);
     sink_params_.Reset(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-        media::CHANNEL_LAYOUT_STEREO, 2, 0, 44100, 16,
+        media::CHANNEL_LAYOUT_STEREO, 2, 44100, 16,
         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
     const int length =
         source_params_.frames_per_buffer() * source_params_.channels();
     source_data_.reset(new int16[length]);
     sink_bus_ = media::AudioBus::Create(sink_params_);
     MockMediaConstraintFactory constraint_factory;
       scoped_refptr<WebRtcAudioCapturer> capturer(
         WebRtcAudioCapturer::CreateCapturer(
             -1, StreamDeviceInfo(),
             constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
@@ skipping 92 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