Add back video_replay. Disappeared in the gn conversion.

webrtc/video/replay.cc

 /*
  *  Copyright (c) 2014 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 <stdio.h>
 
 #include <map>
 #include <memory>
 #include <sstream>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/run_test.h"
@@ skipping 173 matching lines @@
  private:
   FILE* file_;
 };
 
 void RtpReplay() {
   std::unique_ptr<test::VideoRenderer> playback_video(
       test::VideoRenderer::Create("Playback Video", 640, 480));
   FileRenderPassthrough file_passthrough(flags::OutBase(),
                                          playback_video.get());
 
-  std::unique_ptr<Call> call(Call::Create(Call::Config()));
+  webrtc::RtcEventLogNullImpl event_log;
+  std::unique_ptr<Call> call(Call::Create(Call::Config(&event_log)));
 
   test::NullTransport transport;
   VideoReceiveStream::Config receive_config(&transport);
   receive_config.rtp.remote_ssrc = flags::Ssrc();
   receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
   receive_config.rtp.ulpfec.ulpfec_payload_type = flags::FecPayloadType();
   receive_config.rtp.ulpfec.red_payload_type = flags::RedPayloadType();
   receive_config.rtp.nack.rtp_history_ms = 1000;
   if (flags::TransmissionOffsetId() != -1) {
     receive_config.rtp.extensions.push_back(RtpExtension(
@@ skipping 93 matching lines @@
 }
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   ::testing::InitGoogleTest(&argc, argv);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   webrtc::test::RunTest(webrtc::RtpReplay);
   return 0;
 }