Only turn on the audio processing by default for MEDIA_DEVICE_AUDIO_CAPTURE

content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

 // Copyright 2014 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/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
@@ skipping 17 matching lines @@
 
 }  // namespace
 
 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
  public:
   WebRtcLocalAudioTrackAdapterTest()
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480),
         adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)),
         capturer_(WebRtcAudioCapturer::CreateCapturer(
-            -1, StreamDeviceInfo(), blink::WebMediaConstraints(), NULL)),
+            -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
+            blink::WebMediaConstraints(), NULL)),
         track_(new WebRtcLocalAudioTrack(adapter_, capturer_, NULL)) {}
 
  protected:
   virtual void SetUp() OVERRIDE {
     static_cast<WebRtcLocalAudioSourceProvider*>(
         track_->audio_source_provider())->SetSinkParamsForTesting(params_);
     track_->OnSetFormat(params_);
     EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
   }
 
@@ skipping 25 matching lines @@
 
   // Remove the sink from the webrtc track.
   webrtc_track->RemoveSink(sink.get());
   sink.reset();
 
   // Verify that no more callback gets into the sink.
   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
 }
 
 }  // namespace content