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

webrtc/call/rtc_event_log2rtp_dump.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 <iostream>
 #include <sstream>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/rtp_file_writer.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 {
 
 DEFINE_bool(noaudio,
             false,
             "Excludes audio packets from the converted RTPdump file.");
 DEFINE_bool(novideo,
             false,
             "Excludes video packets from the converted RTPdump file.");
@@ skipping 161 matching lines @@
       } else {
         std::cout << "Skipping malformed event." << std::endl;
       }
     }
   }
   std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
             << " RTP packets and " << rtcp_counter << " RTCP packets to the "
             << "output file." << std::endl;
   return 0;
 }