Split webrtc/video into webrtc/{audio,call,video}.

webrtc/call/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.
  */
 
 #ifdef ENABLE_RTC_EVENT_LOG
 
 #include <stdio.h>
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
+#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
-#include "webrtc/video/rtc_event_log.h"
 
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/video/rtc_event_log.pb.h"
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
 #else
-#include "webrtc/video/rtc_event_log.pb.h"
+#include "webrtc/call/rtc_event_log.pb.h"
 #endif
 
 namespace webrtc {
 
 namespace {
 
 const RTPExtensionType kExtensionTypes[] = {
     RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
     RTPExtensionType::kRtpExtensionAudioLevel,
     RTPExtensionType::kRtpExtensionAbsoluteSendTime,
     RTPExtensionType::kRtpExtensionVideoRotation,
     RTPExtensionType::kRtpExtensionTransportSequenceNumber};
 const char* kExtensionNames[] = {RtpExtension::kTOffset,
                                  RtpExtension::kAudioLevel,
                                  RtpExtension::kAbsSendTime,
                                  RtpExtension::kVideoRotation,
                                  RtpExtension::kTransportSequenceNumber};
 const size_t kNumExtensions = 5;
 
-}  // namepsace
+}  // namespace
 
 // TODO(terelius): Place this definition with other parsing functions?
 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
   switch (media_type) {
     case rtclog::MediaType::ANY:
       return MediaType::ANY;
     case rtclog::MediaType::AUDIO:
       return MediaType::AUDIO;
     case rtclog::MediaType::VIDEO:
       return MediaType::VIDEO;
@@ skipping 481 matching lines @@
                             extensions,       // Bit vector choosing extensions
                             csrcs_count,      // Number of contributing sources
                             rand());
     }
   }
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG