Fixing issue with default stream upon setting 2nd remote description.

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 1976 matching lines @@
 // description doesn't contain any streams but does support MSID.
 TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
   EXPECT_EQ(0u, observer_.remote_streams()->count());
 }
 
+// This tests that when setting a new description, the old default tracks are
+// not destroyed and recreated.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
+TEST_F(PeerConnectionInterfaceTest, DefaultTracksNotDestroyedAndRecreated) {
+  FakeConstraints constraints;
+  constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
+                           true);
+  CreatePeerConnection(&constraints);
+  CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
+
+  ASSERT_EQ(1u, observer_.remote_streams()->count());
+  MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
+  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
+
+  // Set the track to "disabled", then set a new description and ensure the
+  // track is still disabled, which ensures it hasn't been recreated.
+  remote_stream->GetAudioTracks()[0]->set_enabled(false);
+  CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
+  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
+  EXPECT_FALSE(remote_stream->GetAudioTracks()[0]->enabled());
+}
+
 // This tests that a default MediaStream is not created if a remote session
 // description is updated to not have any MediaStreams.
 TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
   FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithStream1);
   rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1));
   EXPECT_TRUE(
@@ skipping 318 matching lines @@
   FakeConstraints updated_answer_c;
   answer_c.SetMandatoryReceiveAudio(false);
   answer_c.SetMandatoryReceiveVideo(false);
 
   cricket::MediaSessionOptions updated_answer_options;
   EXPECT_TRUE(
       ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
   EXPECT_TRUE(updated_answer_options.has_audio());
   EXPECT_TRUE(updated_answer_options.has_video());
 }