Remove deprecated functions.

talk/app/webrtc/peerconnectioninterface_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 1083 matching lines @@
 // the answer as a local description.
 TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
   AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
   SessionDescriptionInterface* desc =
       webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
-                                       webrtc::kFireFoxSdpOffer);
+                                       webrtc::kFireFoxSdpOffer, nullptr);
   EXPECT_TRUE(DoSetSessionDescription(desc, false));
   CreateAnswerAsLocalDescription();
   ASSERT_TRUE(pc_->local_description() != NULL);
   ASSERT_TRUE(pc_->remote_description() != NULL);
 
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(pc_->local_description()->description());
   ASSERT_TRUE(content != NULL);
   EXPECT_FALSE(content->rejected);
 
@@ skipping 12 matching lines @@
 // Test that we can create an audio only offer and receive an answer with a
 // limited set of audio codecs and receive an updated offer with more audio
 // codecs, where the added codecs are not supported.
 TEST_F(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
   CreatePeerConnection();
   AddVoiceStream("audio_label");
   CreateOfferAsLocalDescription();
 
   SessionDescriptionInterface* answer =
       webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
-                                       webrtc::kAudioSdp);
+                                       webrtc::kAudioSdp, nullptr);
   EXPECT_TRUE(DoSetSessionDescription(answer, false));
 
   SessionDescriptionInterface* updated_offer =
       webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
-                                       webrtc::kAudioSdpWithUnsupportedCodecs);
+                                       webrtc::kAudioSdpWithUnsupportedCodecs,
+                                       nullptr);
   EXPECT_TRUE(DoSetSessionDescription(updated_offer, false));
   CreateAnswerAsLocalDescription();
 }
 
 // Test that PeerConnection::Close changes the states to closed and all remote
 // tracks change state to ended.
 TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
   // Initialize a PeerConnection and negotiate local and remote session
   // description.
   InitiateCall();
@@ skipping 64 matching lines @@
                                          sdp, NULL);
   EXPECT_FALSE(DoSetLocalDescription(local_offer));
 }
 
 // Test that GetStats can still be called after PeerConnection::Close.
 TEST_F(PeerConnectionInterfaceTest, CloseAndGetStats) {
   InitiateCall();
   pc_->Close();
   DoGetStats(NULL);
 }