Remove FlexfecConfig and replace with specific struct in VideoSendStream.

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 3828 matching lines @@
 
 void VerifyEmptyUlpfecConfig(const UlpfecConfig& config) {
   EXPECT_EQ(-1, config.ulpfec_payload_type)
       << "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
   EXPECT_EQ(-1, config.red_payload_type)
       << "Enabling ULPFEC requires rtpmap: red negotiation.";
   EXPECT_EQ(-1, config.red_rtx_payload_type)
       << "Enabling RTX in ULPFEC requires rtpmap: rtx negotiation.";
 }
 
-void VerifyEmptyFlexfecConfig(const FlexfecConfig& config) {
-  EXPECT_EQ(-1, config.flexfec_payload_type)
+void VerifyEmptyFlexfecConfig(
+    const VideoSendStream::Config::Rtp::Flexfec& config) {
+  EXPECT_EQ(-1, config.payload_type)
       << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
-  EXPECT_EQ(0U, config.flexfec_ssrc)
+  EXPECT_EQ(0U, config.ssrc)
       << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
   EXPECT_TRUE(config.protected_media_ssrcs.empty())
       << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
 }
 
 TEST_P(EndToEndTest, VerifyDefaultSendConfigParameters) {
   VideoSendStream::Config default_send_config(nullptr);
   EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
       << "Enabling NACK require rtcp-fb: nack negotiation.";
   EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
@@ skipping 202 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