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

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  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 "webrtc/media/engine/fakewebrtccall.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/api/call/audio_sink.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/platform_file.h"
-#include "webrtc/base/gunit.h"
 #include "webrtc/media/base/rtputils.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/gunit.h"
+#include "webrtc/rtc_base/platform_file.h"
 
 namespace cricket {
 FakeAudioSendStream::FakeAudioSendStream(
     int id, const webrtc::AudioSendStream::Config& config)
     : id_(id), config_(config) {
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
 void FakeAudioSendStream::Reconfigure(
     const webrtc::AudioSendStream::Config& config) {
@@ skipping 595 matching lines @@
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {
     last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
   }
 }
 
 }  // namespace cricket