Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/sdk/android/src/jni/peerconnection_jni.cc

 /*
  *  Copyright 2013 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.
  */
 
@@ skipping 30 matching lines @@
 
 #include <limits>
 #include <memory>
 #include <utility>
 
 #include "webrtc/api/mediaconstraintsinterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/videosourceproxy.h"
-#include "webrtc/base/bind.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/event_tracer.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/logsinks.h"
-#include "webrtc/base/messagequeue.h"
-#include "webrtc/base/networkmonitor.h"
-#include "webrtc/base/rtccertificategenerator.h"
-#include "webrtc/base/ssladapter.h"
-#include "webrtc/base/stringutils.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/modules/utility/include/jvm_android.h"
 #include "webrtc/pc/webrtcsdp.h"
+#include "webrtc/rtc_base/bind.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/event_tracer.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/logsinks.h"
+#include "webrtc/rtc_base/messagequeue.h"
+#include "webrtc/rtc_base/networkmonitor.h"
+#include "webrtc/rtc_base/rtccertificategenerator.h"
+#include "webrtc/rtc_base/ssladapter.h"
+#include "webrtc/rtc_base/stringutils.h"
 #include "webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.h"
 // Adding 'nogncheck' to disable the gn include headers check.
 // We don't want to always depend on audio and video related targets.
 #include "webrtc/sdk/android/src/jni/androidvideotracksource.h"  // nogncheck
 #include "webrtc/sdk/android/src/jni/audio_jni.h"
 #include "webrtc/sdk/android/src/jni/classreferenceholder.h"
 #include "webrtc/sdk/android/src/jni/jni_helpers.h"
 #include "webrtc/sdk/android/src/jni/media_jni.h"
 #include "webrtc/sdk/android/src/jni/ownedfactoryandthreads.h"
 #include "webrtc/sdk/android/src/jni/rtcstatscollectorcallbackwrapper.h"
@@ skipping 1048 matching lines @@
 
 JOW(jlong, PeerConnectionFactory_nativeCreatePeerConnectionFactory)
 (JNIEnv* jni,
  jclass,
  jobject joptions,
  jobject jencoder_factory,
  jobject jdecoder_factory) {
   // talk/ assumes pretty widely that the current Thread is ThreadManager'd, but
   // ThreadManager only WrapCurrentThread()s the thread where it is first
   // created.  Since the semantics around when auto-wrapping happens in
-  // webrtc/base/ are convoluted, we simply wrap here to avoid having to think
-  // about ramifications of auto-wrapping there.
+  // webrtc/rtc_base/ are convoluted, we simply wrap here to avoid having to
+  // think about ramifications of auto-wrapping there.
   rtc::ThreadManager::Instance()->WrapCurrentThread();
   webrtc::Trace::CreateTrace();
 
   std::unique_ptr<Thread> network_thread =
       rtc::Thread::CreateWithSocketServer();
   network_thread->SetName("network_thread", nullptr);
   RTC_CHECK(network_thread->Start()) << "Failed to start thread";
 
   std::unique_ptr<Thread> worker_thread = rtc::Thread::Create();
   worker_thread->SetName("worker_thread", nullptr);
@@ skipping 1191 matching lines @@
   return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
       ->inter_tone_gap();
 }
 
 JOW(void, DtmfSender_free)
 (JNIEnv* jni, jclass, jlong j_dtmf_sender_pointer) {
   reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)->Release();
 }
 
 }  // namespace webrtc_jni