Wire up RTP keep-alive in ortc api.

webrtc/ortc/ortcrtpsender_unittest.cc

 /*
  *  Copyright 2017 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 22 matching lines @@
     // the "send" flag is applied to the media engine based on the encoding
     // |active| flag.
     fake_packet_transport_.SetWritable(true);
     fake_media_engine_ = new cricket::FakeMediaEngine();
     // Note: This doesn't need to use fake network classes, since we already
     // use FakePacketTransport.
     auto ortc_factory_result = OrtcFactory::Create(
         nullptr, nullptr, nullptr, nullptr, nullptr,
         std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_));
     ortc_factory_ = ortc_factory_result.MoveValue();
-    RtcpParameters rtcp_parameters;
-    rtcp_parameters.mux = true;
+    RtpTransportParameters parameters;
+    parameters.rtcp.mux = true;
     auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
-        rtcp_parameters, &fake_packet_transport_, nullptr, nullptr);
+        parameters, &fake_packet_transport_, nullptr, nullptr);
     rtp_transport_ = rtp_transport_result.MoveValue();
   }
 
  protected:
   rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
       const std::string& id) {
     return ortc_factory_->CreateAudioTrack(id, nullptr);
   }
 
   rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
@@ skipping 89 matching lines @@
       cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
   auto video_sender = video_sender_result.MoveValue();
   EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
             video_sender->SetTrack(CreateAudioTrack("audio")).type());
 }
 
 // Currently SetTransport isn't supported. When it is, replace this test with a
 // test/tests for it.
 TEST_F(OrtcRtpSenderTest, SetTransportFails) {
   rtc::FakePacketTransport fake_packet_transport("another_transport");
-  RtcpParameters rtcp_parameters;
-  rtcp_parameters.mux = true;
+  RtpTransportParameters parameters;
+  parameters.rtcp.mux = true;
   auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
-      rtcp_parameters, &fake_packet_transport, nullptr, nullptr);
+      parameters, &fake_packet_transport, nullptr, nullptr);
   auto rtp_transport = rtp_transport_result.MoveValue();
 
   auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
                                                       rtp_transport_.get());
   auto sender = sender_result.MoveValue();
   EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
             sender->SetTransport(rtp_transport.get()).type());
 }
 
 TEST_F(OrtcRtpSenderTest, GetTransport) {
@@ skipping 488 matching lines @@
       cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
   auto video_sender_result = ortc_factory_->CreateRtpSender(
       CreateVideoTrack("video"), rtp_transport_.get());
   auto audio_sender = audio_sender_result.MoveValue();
   auto video_sender = video_sender_result.MoveValue();
   EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, audio_sender->GetKind());
   EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, video_sender->GetKind());
 }
 
 }  // namespace webrtc