Delete member VideoReceiveStream::Config::Rtp::ulpfec.

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 220 matching lines @@
     audio_send_config.send_codec_spec =
         rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
             {kAudioSendPayloadType, {"ISAC", 16000, 1}});
     audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
     audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config);
 
     video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     if (fec == FecMode::kOn) {
       video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType;
       video_send_config_.rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
-      video_receive_configs_[0].rtp.ulpfec.red_payload_type = kRedPayloadType;
-      video_receive_configs_[0].rtp.ulpfec.ulpfec_payload_type =
-          kUlpfecPayloadType;
+      video_receive_configs_[0].rtp.red_payload_type = kRedPayloadType;
+      video_receive_configs_[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
     }
     video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
     video_receive_configs_[0].renderer = &observer;
     video_receive_configs_[0].sync_group = kSyncGroup;
 
     AudioReceiveStream::Config audio_recv_config;
     audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
     audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
     audio_recv_config.voe_channel_id = recv_channel_id;
     audio_recv_config.sync_group = kSyncGroup;
@@ skipping 525 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc