Move rtp dump writer from quality test to test transport

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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 <algorithm>
@@ skipping 4600 matching lines @@
   static const int kReceiveDelayMs = 70;
 
   std::unique_ptr<test::DirectTransport> sender_transport;
   std::unique_ptr<test::DirectTransport> receiver_transport;
 
   task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
     FakeNetworkPipe::Config config;
     config.queue_delay_ms = kSendDelayMs;
     CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
     sender_transport = rtc::MakeUnique<test::DirectTransport>(
-        &task_queue_, config, sender_call_.get(), payload_type_map_);
+        &task_queue_, config, sender_call_.get(), payload_type_map_,
+        std::unique_ptr<test::RtpFileWriter>());
     config.queue_delay_ms = kReceiveDelayMs;
     receiver_transport = rtc::MakeUnique<test::DirectTransport>(
-        &task_queue_, config, receiver_call_.get(), payload_type_map_);
+        &task_queue_, config, receiver_call_.get(), payload_type_map_,
+        std::unique_ptr<test::RtpFileWriter>());
     sender_transport->SetReceiver(receiver_call_->Receiver());
     receiver_transport->SetReceiver(sender_call_->Receiver());
 
     CreateSendConfig(1, 0, 0, sender_transport.get());
     CreateMatchingReceiveConfigs(receiver_transport.get());
 
     CreateVideoStreams();
     CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
                                  kDefaultHeight);
     Start();
@@ skipping 381 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc