Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

webrtc/test/call_test.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.
  */
 
@@ skipping 257 matching lines @@
     audio_config.rtcp_send_transport = rtcp_send_transport;
     audio_config.voe_channel_id = voe_recv_.channel_id;
     audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
     audio_config.decoder_factory = decoder_factory_;
     audio_receive_configs_.push_back(audio_config);
   }
 
   // TODO(brandtr): Update this when we support multistream protection.
   RTC_DCHECK(num_flexfec_streams_ <= 1);
   if (num_flexfec_streams_ == 1) {
-    FlexfecReceiveStream::Config config;
+    FlexfecReceiveStream::Config config(rtcp_send_transport);
     config.payload_type = kFlexfecPayloadType;
     config.remote_ssrc = kFlexfecSendSsrc;
     config.protected_media_ssrcs = {kVideoSendSsrcs[0]};
     for (const RtpExtension& extension : video_send_config_.rtp.extensions)
       config.rtp_header_extensions.push_back(extension);
     flexfec_receive_configs_.push_back(config);
   }
 }
 
 void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
@@ skipping 219 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc