Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

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 3870 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyUlpfecConfig(default_receive_config.rtp.ulpfec);
 }
 
 TEST_P(EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) {
-  FlexfecReceiveStream::Config default_receive_config;
+  FlexfecReceiveStream::Config default_receive_config(nullptr);
   EXPECT_EQ(-1, default_receive_config.payload_type)
       << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
   EXPECT_EQ(0U, default_receive_config.remote_ssrc)
       << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
   EXPECT_TRUE(default_receive_config.protected_media_ssrcs.empty())
       << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
 }
 
 TEST_P(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
   static const int kExtensionId = 8;
@@ skipping 162 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