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

webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.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/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
 
 #include <assert.h>
 #include <string.h>
 #include <iostream>
 #include <limits>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.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 test {
 
 namespace {
 
 const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) {
   if (!event.has_type() || event.type() != rtclog::Event::RTP_EVENT)
     return nullptr;
@@ skipping 80 matching lines @@
 RtcEventLogSource::RtcEventLogSource()
     : PacketSource(), parser_(RtpHeaderParser::Create()) {}
 
 bool RtcEventLogSource::OpenFile(const std::string& file_name) {
   event_log_.reset(new rtclog::EventStream());
   return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get());
 }
 
 }  // namespace test
 }  // namespace webrtc