Add FlexFEC to CallTest.

webrtc/call/call_perf_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.
  */
 
@@ skipping 195 matching lines @@
                                          MediaType::VIDEO);
   video_send_transport.SetReceiver(&video_receiver);
 
   test::PacketTransport receive_transport(
       receiver_call_.get(), &observer, test::PacketTransport::kReceiver,
       FakeNetworkPipe::Config());
   receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
-  CreateSendConfig(1, 0, &video_send_transport);
+  CreateSendConfig(1, 0, 0, &video_send_transport);
   CreateMatchingReceiveConfigs(&receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
   audio_send_config.rtp.ssrc = kAudioSendSsrc;
   audio_send_config.send_codec_spec.codec_inst =
       CodecInst{103, "ISAC", 16000, 480, 1, 32000};
   AudioSendStream* audio_send_stream =
       sender_call_->CreateAudioSendStream(audio_send_config);
 
@@ skipping 499 matching lines @@
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc