Roll WebRTC 11523:11548, Libjingle 11522:11545

content/renderer/media/media_stream_audio_processor_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 <stddef.h>
 #include <stdint.h>
 
 #include <vector>
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
 #include "base/macros.h"
 #include "base/memory/aligned_memory.h"
 #include "base/path_service.h"
 #include "base/time/time.h"
 #include "build/build_config.h"
 #include "content/public/common/media_stream_request.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
-#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 using ::testing::AtLeast;
 using ::testing::Return;
 
 using media::AudioParameters;
 
 namespace webrtc {
 
@@ skipping 556 matching lines @@
   ProcessDataAndVerifyFormat(audio_processor.get(),
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content