Remove FlexfecConfig and replace with specific struct in VideoSendStream.

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 213 matching lines @@
   if (num_audio_streams > 0) {
     audio_send_config_ = AudioSendStream::Config(send_transport);
     audio_send_config_.voe_channel_id = voe_send_.channel_id;
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
     audio_send_config_.send_codec_spec.codec_inst =
         CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
   }
 
   // TODO(brandtr): Update this when we support multistream protection.
   if (num_flexfec_streams > 0) {
-    video_send_config_.rtp.flexfec.flexfec_payload_type = kFlexfecPayloadType;
-    video_send_config_.rtp.flexfec.flexfec_ssrc = kFlexfecSendSsrc;
+    video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
+    video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
     video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
   }
 }
 
 void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
   RTC_DCHECK(video_receive_configs_.empty());
   RTC_DCHECK(allocated_decoders_.empty());
   if (num_video_streams_ > 0) {
     RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
     VideoReceiveStream::Config video_config(rtcp_send_transport);
@@ skipping 262 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc