Moved RtcEventLog files from call/ to logging/ (new top level dir)

webrtc/logging/rtc_event_log/rtc_event_log_unittest.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 <map>
 #include <memory>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
-#include "webrtc/call/rtc_event_log_parser.h"
-#include "webrtc/call/rtc_event_log_unittest_helper.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
-#include "webrtc/call/rtc_event_log.pb.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 
 namespace webrtc {
 
 namespace {
 
 const RTPExtensionType kExtensionTypes[] = {
     RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
     RTPExtensionType::kRtpExtensionAudioLevel,
     RTPExtensionType::kRtpExtensionAbsoluteSendTime,
@@ skipping 402 matching lines @@
                                          MediaType::VIDEO, rtcp_packet.data(),
                                          rtcp_packet.size());
 
   RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3);
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
 
 }  // namespace webrtc