base->rtc_base: Update .c, .mm and .java files.

webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <algorithm>
 #include <limits>
 #include <list>
 #include <memory>
 #include <numeric>
 #include <string>
 #include <vector>
 
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/format_macros.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/include/mock_audio_transport.h"
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
+#include "webrtc/rtc_base/arraysize.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/format_macros.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 #import "webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
 #import "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
 
 using std::cout;
 using std::endl;
@@ skipping 429 matching lines @@
   size_t rec_count_;
   AudioStreamInterface* audio_stream_;
 };
 
 // AudioDeviceTest test fixture.
 class AudioDeviceTest : public ::testing::Test {
  protected:
   AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
     old_sev_ = rtc::LogMessage::GetLogToDebug();
     // Set suitable logging level here. Change to rtc::LS_INFO for more verbose
-    // output. See webrtc/base/logging.h for complete list of options.
+    // output. See webrtc/rtc_base/logging.h for complete list of options.
     rtc::LogMessage::LogToDebug(rtc::LS_INFO);
     // Add extra logging fields here (timestamps and thread id).
     // rtc::LogMessage::LogTimestamps();
     rtc::LogMessage::LogThreads();
     // Creates an audio device using a default audio layer.
     audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
     EXPECT_NE(audio_device_.get(), nullptr);
     EXPECT_EQ(0, audio_device_->Init());
     EXPECT_EQ(0,
               audio_device()->GetPlayoutAudioParameters(&playout_parameters_));
@@ skipping 382 matching lines @@
   // Wait for notification to propagate.
   rtc::MessageQueueManager::ProcessAllMessageQueues();
   EXPECT_TRUE(audio_device->is_interrupted_);
 
   audio_device->Init();
   audio_device->InitPlayout();
   EXPECT_FALSE(audio_device->is_interrupted_);
 }
 
 }  // namespace webrtc