Switch LiveAudio to source provider solution.

content/renderer/media/webrtc_audio_device_unittest.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 <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 101 matching lines @@
       WebRtcAudioCapturer::CreateCapturer());
 
   media::AudioHardwareConfig* hardware_config =
       RenderThreadImpl::current()->GetAudioHardwareConfig();
 
   // Use native capture sample rate and channel configuration to get some
   // action in this test.
   int sample_rate = hardware_config->GetInputSampleRate();
   media::ChannelLayout channel_layout =
       hardware_config->GetInputChannelLayout();
-  if (!capturer->Initialize(kRenderViewId, channel_layout, sample_rate, 1,
+  if (!capturer->Initialize(kRenderViewId, channel_layout, sample_rate, 0, 1,
                             media::AudioManagerBase::kDefaultDeviceId)) {
     return false;
   }
 
   // Add the capturer to the WebRtcAudioDeviceImpl.
   webrtc_audio_device->AddAudioCapturer(capturer);
 
   return true;
 }
 
 // Create and start a local audio track. Starting the audio track will connect
 // the audio track to the capturer and also start the source of the capturer.
 // Also, connect the sink to the audio track.
 scoped_refptr<WebRtcLocalAudioTrack>
 CreateAndStartLocalAudioTrack(WebRtcAudioCapturer* capturer,
                               WebRtcAudioCapturerSink* sink) {
   scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
-      WebRtcLocalAudioTrack::Create(std::string(), capturer, NULL, NULL));
+      WebRtcLocalAudioTrack::Create(std::string(), capturer, NULL, NULL, NULL));
   local_audio_track->AddSink(sink);
   local_audio_track->Start();
   return local_audio_track;
 }
 
 class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
  public:
   explicit WebRTCMediaProcessImpl(base::WaitableEvent* event)
       : event_(event),
         channel_id_(-1),
@@ skipping 844 matching lines @@
 #endif
 
 TEST_F(MAYBE_WebRTCAudioDeviceTest,
        MAYBE_WebRtcLoopbackTimeWithSignalProcessing) {
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content