Fix the issue of setting local description with datachannel only and max-bundle.

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 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 1744 matching lines @@
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                  true);
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
   receiving_client()->RemoveSdesCryptoFromReceivedSdp(true);
   LocalP2PTest();
   VerifyRenderedSize(640, 480);
 }
 
+// This test verifies that the negotiation will success with data channel only

[Taylor Brandstetter, 2016/11/02 23:10:22]

"will success" -> "will succeed"

+// in max-bundle mode.
+TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) {
+  webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
+  rtc_config.bundle_policy =
+      webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
+  ASSERT_TRUE(CreateTestClients(rtc_config, rtc_config));
+  initializing_client()->CreateDataChannel();
+  initializing_client()->AddMediaStream(false /*No audio*/, false /*No video*/);

[Taylor Brandstetter, 2016/11/02 23:10:22]

Just curious, why is adding a media stream necessary? Is it due to the weird
test infrastructure? May be worth leaving a comment explaining.

+  initializing_client()->Negotiate();
+}
+
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive video.
 TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->SetReceiveAudioVideo(false, true);
   LocalP2PTest();
 }
 
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive audio.
@@ skipping 985 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace