Roll WebRTC 11523:11548, Libjingle 11522:11545

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 <stddef.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_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"
+#include "third_party/webrtc/api/mediastreaminterface.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 
 namespace content {
 
 namespace {
 
 class MockWebRtcAudioSink : public webrtc::AudioTrackSinkInterface {
  public:
@@ skipping 69 matching lines @@
 }
 
 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
   webrtc::AudioTrackInterface* webrtc_track =
       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
   int signal_level = 0;
   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
 }
 
 }  // namespace content